[要約] RFC 3666は、SIPとPSTNの通話フローに関する情報を提供する。その目的は、SIPとPSTNの相互運用性を向上させ、通信ネットワークの設計や実装に役立つことである。
Network Working Group A. Johnston
Request for Comments: 3666 MCI
BCP: 76 S. Donovan
Category: Best Current Practice R. Sparks
C. Cunningham
dynamicsoft
K. Summers
Sonus
December 2003
Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows
セッション開始プロトコル（SIP）パブリックスイッチ付き電話ネットワーク（PSTN）コールフロー
Status of this Memo
本文書の位置付け
This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. Distribution of this memo is unlimited.
このドキュメントは、インターネットコミュニティのインターネットの最良のプラクティスを指定し、改善のための議論と提案を要求します。このメモの配布は無制限です。
Copyright Notice
著作権表示
Copyright (C) The Internet Society (2003). All Rights Reserved.
Copyright（c）The Internet Society（2003）。無断転載を禁じます。
Abstract
概要
This document contains best current practice examples of Session Initiation Protocol (SIP) call flows showing interworking with the Public Switched Telephone Network (PSTN). Elements in these call flows include SIP User Agents, SIP Proxy Servers, and PSTN Gateways. Scenarios include SIP to PSTN, PSTN to SIP, and PSTN to PSTN via SIP. PSTN telephony protocols are illustrated using ISDN (Integrated Services Digital Network), ISUP (ISDN User Part), and FGB (Feature Group B) circuit associated signaling. PSTN calls are illustrated using global telephone numbers from the PSTN and private extensions served on by a PBX (Private Branch Exchange). Call flow diagrams and message details are shown.
このドキュメントには、セッション開始プロトコル（SIP）コールフローの最良の現在の練習例が含まれています。これらのコールフローの要素には、SIPユーザーエージェント、SIPプロキシサーバー、PSTNゲートウェイが含まれます。シナリオには、PSTNへのSIP、SIPへのPSTN、SIPを介してPSTNにPSTNが含まれます。PSTNテレフォニープロトコルは、ISDN（Integrated Services Digital Network）、ISUP（ISDNユーザーパーツ）、およびFGB（特徴グループB）回路関連シグナルを使用して示されています。PSTNコールは、PSN（Private Branch Exchange）が提供するPSTNおよびプライベートエクステンションのグローバルな電話番号を使用して示されています。コールフロー図とメッセージの詳細が表示されます。
Table of Contents
目次
1. Overview..................................................... 2
1.1. General Assumptions.................................... 3
1.2. Legend for Message Flows............................... 4
1.3. SIP Protocol Assumptions............................... 5
2. SIP to PSTN Dialing.......................................... 6
2.1. Successful SIP to ISUP PSTN call....................... 7
2.2. Successful SIP to ISDN PBX call........................ 15
2.3. Successful SIP to ISUP PSTN call with overflow......... 23
2.4. Session established using ENUM Query................... 32
2.5. Unsuccessful SIP to PSTN call: Treatment from PSTN..... 38
2.6. Unsuccessful SIP to PSTN: REL w/Cause from PSTN........ 45
2.7. Unsuccessful SIP to PSTN: ANM Timeout.................. 49
3. PSTN to SIP Dialing.......................................... 54
3.1. Successful PSTN to SIP call............................ 55
3.2. Successful PSTN to SIP call, Fast Answer............... 62
3.3. Successful PBX to SIP call............................. 68
3.4. Unsuccessful PSTN to SIP REL, SIP error mapped to REL.. 74
3.5. Unsuccessful PSTN to SIP REL, SIP busy mapped to REL... 76
3.6. Unsuccessful PSTN->SIP, SIP error interworking to tones 80
3.7. Unsuccessful PSTN->SIP, ACM timeout.................... 84
3.8. Unsuccessful PSTN->SIP, ACM timeout, stateless Proxy... 88
3.9. Unsuccessful PSTN->SIP, Caller Abandonment............. 91
4. PSTN to PSTN Dialing via SIP Network......................... 96
4.1. Successful ISUP PSTN to ISUP PSTN call................. 97
4.2. Successful FGB PBX to ISDN PBX call with overflow...... 105
5. Security Considerations...................................... 113
6. References................................................... 115
6.1. Normative References................................... 115
6.2. Informative References................................. 115
7. Acknowledgments.............................................. 116
8. Intellectual Property Statement.............................. 116
9. Authors' Addresses........................................... 117
10. Full Copyright Statement..................................... 118
The call flows shown in this document were developed in the design of a SIP IP communications network. They represent an example of a minimum set of functionality.
このドキュメントに示されているコールフローは、SIP IP通信ネットワークの設計で開発されました。これらは、最小の機能セットの例を表しています。
It is the hope of the authors that this document will be useful for SIP implementers, designers, and protocol researchers alike and will help further the goal of a standard implementation of RFC 3261 [2]. These flows represent carefully checked and working group reviewed scenarios of the most common SIP/PSTN interworking examples as a companion to the specifications.
著者の希望は、このドキュメントがSIP実装者、設計者、およびプロトコル研究者にとっても役立ち、RFC 3261の標準的な実装の目標を促進するのに役立つことを希望します[2]。これらのフローは、慎重にチェックされ、ワーキンググループが最も一般的なSIP/PSTNインターワーキングの例のシナリオを仕様のコンパニオンとしてレビューしたことを表しています。
These call flows are based on the current version 2.0 of SIP in RFC 3261 [2] with SDP usage described in RFC 3264 [3]. Other RFCs also comprise the SIP standard but are not used in this set of basic call flows. The SIP/ISUP mapping is based on RFC 3398 [4].
これらのコールフローは、RFC 3261 [2]のSIPの現在のバージョン2.0に基づいており、RFC 3264 [3]で説明されているSDP使用法。他のRFCもSIP標準を構成しますが、この一連の基本的なコールフローでは使用されません。SIP/ISUPマッピングは、RFC 3398 [4]に基づいています。
Various PSTN signaling protocols are illustrated in this document: ISDN (Integrated Services Digital Network), ISUP (ISDN User Part) and FGB (Feature Group B) circuit associated signaling. This document shows mainly ANSI ISUP due to its practical origins. However, as used in this document, the usage is virtually identical to the ITU-T International ISUP used as the reference in [4].
このドキュメントには、さまざまなPSTNシグナル伝達プロトコルが示されています。ISDN（Integrated Services Digital Network）、ISUP（ISDNユーザーパーツ）、FGB（FeatureグループB）回路関連シグナル伝達。このドキュメントは、主にその実用的な起源のためにANSI ISUPを示しています。ただし、このドキュメントで使用されているように、使用法は[4]の参照として使用されるITU-T International ISUPとほぼ同じです。
Basic SIP call flow examples are contained in a companion document, RFC 3665 [10].
基本的なSIPコールフローの例は、コンパニオンドキュメントRFC 3665 [10]に含まれています。
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14, RFC 2119 [1].
「必須」、「そうしない」、「必須」、「必要」、「「しない」、「そうでない」、「そうではない」、「そうでない」、「推奨」、「5月」、および「オプション」は、BCP 14、RFC 2119 [1]に記載されているように解釈される。
A number of architecture, network, and protocol assumptions underlie the call flows in this document. Note that these assumptions are not requirements. They are outlined in this section so that they may be taken into consideration and to aid in the understanding of the call flow examples.
このドキュメントのコールフローの根底にある多くのアーキテクチャ、ネットワーク、およびプロトコルの仮定があります。これらの仮定は要件ではないことに注意してください。これらは、このセクションで概説されているため、考慮され、コールフローの例の理解を支援することができます。
The authentication of SIP User Agents in these example call flows is performed using HTTP Digest as defined in [3] and [5].
これらの例コールフローのSIPユーザーエージェントの認証は、[3]および[5]で定義されているHTTPダイジェストを使用して実行されます。
Some Proxy Servers in these call flows insert Record-Route headers into requests to ensure that they are in the signaling path for future message exchanges.
これらのコールフローの一部のプロキシサーバーは、レコードルートヘッダーをリクエストに挿入して、将来のメッセージ交換の信号パスにいることを確認します。
These flows show TLS, TCP, and UDP for transport. SCTP could also be used. See the discussion in RFC 3261 [2] for details on the transport issues for SIP.
これらのフローは、輸送用のTLS、TCP、およびUDPを示しています。SCTPも使用できます。SIPの輸送問題の詳細については、RFC 3261 [2]のディスカッションを参照してください。
The SIP Proxy Server has access to a Location Service and other databases. Information present in the Request-URI and the context (From header) is sufficient to determine to which proxy or gateway the message should be routed. In most cases, a primary and secondary route will be determined in case of a Proxy or Gateway failure downstream.
SIPプロキシサーバーには、ロケーションサービスやその他のデータベースにアクセスできます。リクエスト-URIおよびコンテキスト（ヘッダーから）に存在する情報は、メッセージをルーティングするプロキシまたはゲートウェイを決定するのに十分です。ほとんどの場合、下流のプロキシまたはゲートウェイの障害が発生した場合に、一次および二次ルートが決定されます。
Gateways provide tones (ringing, busy, etc) and announcements to the PSTN side based on SIP response messages, or pass along audio in-band tones (ringing, busy tone, etc.) in an early media stream to the SIP side.
ゲートウェイは、SIP応答メッセージに基づいてトーン（リンギング、ビジーなど）とPSTN側へのアナウンスを提供するか、SIP側への初期のメディアストリームで帯域内のトーン（リンギング、ビジートーンなど）を渡します。
The interactions between the Proxy and Gateway can be summarized as follows:
プロキシとゲートウェイの間の相互作用は、次のように要約できます。
- The SIP Proxy Server performs digit analysis and lookup and locates the correct gateway.
- SIPプロキシサーバーは、数字分析とルックアップを実行し、正しいゲートウェイを見つけます。
- The SIP Proxy Server performs gateway location based on primary and secondary routing.
- SIPプロキシサーバーは、プライマリおよびセカンダリルーティングに基づいてゲートウェイの場所を実行します。
Telephone numbers are usually represented as SIP URIs. Note that an alternative is the use of the tel URI [6].
通常、電話番号はSIP URIとして表されます。別の方法は、Tel URIの使用であることに注意してください[6]。
This document shows typical examples of SIP/ISUP interworking. Although in the spirit of the SIP-T framework [7], these examples do not represent a complete implementation of the framework. The examples here represent more of a minimal set of examples for very basic SIP to ISUP interworking, rather than the more complex goal of ISUP transparency. In particular, there are NO examples of encapsulated ISUP in this document. If present, these messages would show S/MIME encryption due to the sensitive nature of this information, as discussed in the SIP-T Framework security considerations section. (Note - RFC 3204 [8] contains an example of an INVITE with encapsulated ISUP.) See the Security Considerations section for a more detailed discussion on the security of these call flows.
このドキュメントは、SIP/ISUPインターワーキングの典型的な例を示しています。SIP-Tフレームワーク[7]の精神では、これらの例はフレームワークの完全な実装を表していません。ここでの例は、ISUPの透明性のより複雑な目標ではなく、非常に基本的なSIPのISUPインターワーキングの最小限の例のセットをより多く表しています。特に、このドキュメントにはカプセル化されたISUPの例はありません。存在する場合、これらのメッセージは、SIP-Tフレームワークセキュリティに関する考慮事項セクションで説明されているように、この情報の敏感な性質により、S/MIME暗号化を表示します。（注-RFC 3204 [8]には、カプセル化されたISUPを含む招待の例が含まれています。）これらのコールフローのセキュリティに関する詳細な議論については、セキュリティ上の考慮事項セクションを参照してください。
In ISUP, the Calling Party Number is abbreviated as CgPN and the Called Party Number is abbreviated as CdPN. Other abbreviations include Numbering Plan Indicator (NPI) and Nature of Address (NOA).
ISUPでは、召し当事者番号はCGPNとして略され、呼び出されたパーティー番号はCDPNとして短縮されます。その他の略語には、番号付け計画インジケーター（NPI）と住所の性質（NOA）が含まれます。
Dashed lines (---) represent signaling messages that are mandatory to the call scenario. These messages can be SIP or PSTN signaling. The arrow indicates the direction of message flow.
破線（---）は、コールシナリオに必須のシグナルメッセージを表します。これらのメッセージは、SIPまたはPSTNシグナル伝達にすることができます。矢印は、メッセージフローの方向を示します。
Double dashed lines (===) represent media paths between network elements.
ダブルダッシュライン（===）は、ネットワーク要素間のメディアパスを表します。
Messages with parentheses around their name represent optional messages.
名前の周りに括弧が付いたメッセージは、オプションのメッセージを表します。
Messages are identified in the Figures as F1, F2, etc. This references the message details in the list that follows the Figure. Comments in the message details are shown in the following form:
メッセージは、図でF1、F2などとして識別されます。これは、図に次のようなメッセージの詳細を参照しています。メッセージの詳細のコメントは、次の形式で表示されます。
/* Comments. */
This document does not prescribe the flows precisely as they are shown, but rather the flows illustrate the principles for best practice. They are best practices usages (orderings, syntax, selection of features for the purpose, handling of error) of SIP methods, headers and parameters. IMPORTANT: The exact flows here must not be copied as is by an implementer due to specific incorrect characteristics that were introduced into the document for convenience and are listed below. To sum up, the SIP/PSTN call flows represent well-reviewed examples of SIP usage, which are best common practice according to IETF consensus.
このドキュメントでは、フローが表示されている間、フローを正確に規定していませんが、むしろフローがベストプラクティスの原則を示しています。これらは、SIPメソッド、ヘッダー、パラメーターのベストプラクティスの使用（注文、構文、目的のための機能の選択、エラーの処理）です。重要：ここでの正確なフローは、利便性のためにドキュメントに導入され、以下にリストされている特定の誤った特性のために、実装者がそうであるようにコピーしてはなりません。要約すると、SIP/PSTNコールフローは、IETFのコンセンサスに従って最も一般的な慣行であるSIP使用のよくレビューされた例を表しています。
For simplicity in reading and editing the document, there are a number of differences between some of the examples and actual SIP messages. For example, the SIP Digest responses are not actual MD5 encodings. Call-IDs are often repeated, and CSeq counts often begin at 1. Header fields are usually shown in the same order. Usually only the minimum required header field set is shown, others that would normally be present, such as Accept, Supported, Allow, etc. are not shown.
ドキュメントの読み取りと編集を簡単にするために、いくつかの例と実際のSIPメッセージの間には多くの違いがあります。たとえば、SIPダイジェスト応答は実際のMD5エンコーディングではありません。Call-IDはしばしば繰り返され、CSEQカウントはしばしば1で始まります。ヘッダーフィールドは通常同じ順序で表示されます。通常、必要な最小ヘッダーフィールドセットのみが表示され、通常は受け入れ、サポート、許可など、通常存在するものは表示されません。
Actors:
俳優：
Element Display Name URI IP Address
------- ------------ --- ----------
User Agent Alice sip:alice@a.example.com 192.0.2.101
User Agent Bob sip:bob@b.example.com 192.0.2.200
Proxy Server sip:ss1.a.example.com 192.0.2.111
User Agent (Gateway) sip:gw1.a.example.com 192.0.2.201
User Agent (Gateway) sip:gw2.a.example.com 192.0.2.202
User Agent (Gateway) sip:gw3.a.example.com 192.0.2.203
User Agent (Gateway) sip:ngw1.a.example.com 192.0.2.103
User Agent (Gateway) sip:ngw2.a.example.com 192.0.2.102
Note that NGW 1 and NGW 2 also have device URIs (Contacts) of sip:ngw1@a.example.com and sip:ngw2@a.example.com which resolve to the Proxy Server sip:ss1.wcom.com using DNS SRV records.
NGW 1およびNGW 2には、sip：ngw1@a.example.comおよびsip：ngw2@a.example.comのsip：ngw1@a.example.comのデバイスURI（連絡先）もあります。記録。
In the following scenarios, Alice (sip:alice@a.example.com) is a SIP phone or other SIP-enabled device. Bob is reachable via the PSTN at global telephone number +19725552222. Alice places a call to Bob through a Proxy Server, Proxy 1, and a Network Gateway. In other scenarios, Alice places calls to Carol, who is served via a PBX (Private Branch Exchange) and is identified by a private extension 444-3333, or global number +1-918-555-3333. Note that Alice uses his/her global telephone number +1-314-555-1111 in the From header in the INVITE messages. This then gives the Gateway the option of using this header to populate the calling party identification field in subsequent signaling. Left open is the issue of how the Gateway can determine the accuracy of the telephone number which is necessary before passing it as a valid calling party number in the PSTN.
次のシナリオでは、Alice（sip：alice@a.example.com）は、SIP電話またはその他のSIP対応デバイスです。ボブは、グローバル電話番号19725552222のPSTNを介して到達可能です。アリスは、プロキシサーバー、プロキシ1、およびネットワークゲートウェイを介してボブに電話をかけます。他のシナリオでは、アリスはPBX（プライベートブランチエクスチェンジ）を介して提供され、プライベートエクステンション444-3333、またはグローバル番号1-918-555-3333によって識別されるキャロルに呼びかけます。アリスは、招待メッセージのHeaderでグローバルな電話番号1-314-555-1111を使用していることに注意してください。これにより、ゲートウェイにこのヘッダーを使用して、後続のシグナリングで呼び出しパーティーの識別フィールドに入力するオプションが得られます。左開いたのは、ゲートウェイがPSTNの有効な呼び出しパーティー番号として渡す前に必要な電話番号の正確性をどのように決定できるかという問題です。
In these scenarios, Alice is a SIP phone or other SIP-enabled device. Alice places a call to Bob in the PSTN or Carol on a PBX through a Proxy Server and a Gateway.
これらのシナリオでは、アリスはSIP電話またはその他のSIP対応デバイスです。アリスは、プロキシサーバーとゲートウェイを介してPBXでPSTNまたはキャロルでボブに電話をかけます。
In the failure scenarios, the call does not complete. In some cases however, a media stream is still setup. This is due to the fact that some failures in dialing to the PSTN result in in-band tones (busy, reorder tones or announcements - "The number you have dialed has changed. The new number is..."). The 183 Session Progress response containing SDP media information is used to setup this early media path so that the caller Alice knows the final disposition of the call.
障害シナリオでは、呼び出しが完了しません。ただし、場合によっては、メディアストリームがまだセットアップされています。これは、PSTNへのダイヤルに障害が発生したため、インバンドトーン（ビジー、再注文のトーン、またはアナウンスが発表されるためです。SDPメディア情報を含む183セッションの進行状況応答を使用して、この初期のメディアパスをセットアップして、発信者アリスがコールの最終処分を知っているようにします。
The media stream is either terminated by the caller after the tone or announcement has been heard and understood, or by the Gateway after a timer expires.
メディアストリームは、トーンまたは発表が聞かれ、理解された後、発信者によって終了するか、タイマーが期限切れになった後のゲートウェイによって終了します。
In other failure scenarios, a SS7 Release with Cause Code is mapped to a SIP response. In these scenarios, the early media path is not used, but the actual failure code is conveyed to the caller by the SIP User Agent Client.
他の障害シナリオでは、原因コードを使用したSS7リリースがSIP応答にマッピングされます。これらのシナリオでは、初期のメディアパスは使用されませんが、実際の障害コードはSIPユーザーエージェントクライアントによって発信者に伝えられます。
Alice Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | ACM F6 |
| | 183 F7 |<---------------|
| 183 F8 |<---------------| |
|<---------------| | |
| Both Way RTP Media | One Way Voice |
|<===============================>|<===============|
| | | ANM F9 |
| | 200 F10 |<---------------|
| 200 F11 |<---------------| |
|<---------------| | |
| ACK F12 | | |
|--------------->| ACK F13 | |
| |--------------->| |
| Both Way RTP Media | Both Way Voice |
|<===============================>|<==============>|
| BYE F14 | | |
|--------------->| BYE F15 | |
| |--------------->| |
| | 200 F16 | |
| 200 F17 |<---------------| REL F18 |
|<---------------| |--------------->|
| | | RLC F19 |
| | |<---------------|
| | | |
Alice dials the globalized E.164 number +19725552222 to reach Bob. Note that A might have only dialed the last 7 digits, or some other dialing plan. It is assumed that the SIP User Agent Client converts the digits into a global number and puts them into a SIP URI. Note that tel URIs could be used instead of SIP URIs.
アリスは、グローバル化E.164番号1972552222にダイヤルしてボブに到達します。Aは、最後の7桁、または他のダイヤル計画のみをダイヤルしている可能性があることに注意してください。SIPユーザーエージェントクライアントは、数字をグローバル番号に変換し、それらをSIP URIに配置すると想定されています。SIP URIの代わりにTel Urisを使用できることに注意してください。
Alice could use either their SIP address (sip:alice@a.example.com) or SIP telephone number (sip:+13145551111@ss1.a.example.com;user=phone) in the From header. In this example, the telephone number is included, and it is shown as being passed as calling party identification through the Network Gateway (NGW 1) to Bob (F5). Note that for this number to be passed into the SS7 network, it would have to be somehow verified for accuracy.
アリスは、SIPアドレス（sip：alice@a.example.com）またはSIP電話番号（sip：1314551111@ss1.a.example.com; user =電話）をFrom Headerで使用できます。この例では、電話番号が含まれており、ネットワークゲートウェイ（NGW 1）を介してボブ（F5）に当事者の識別を呼び出すものとして渡されていることが示されています。この数値をSS7ネットワークに渡すには、正確さのために何らかの形で検証する必要があることに注意してください。
In this scenario, Bob answers the call, then Alice disconnects the call. Signaling between NGW 1 and Bob's telephone switch is ANSI ISUP. For the details of SIP to ISUP mapping, refer to [4].
このシナリオでは、ボブは電話に応答し、アリスは電話を切断します。NGW 1とボブの電話スイッチの間のシグナリングは、ANSI ISUPです。SIPからISUPマッピングの詳細については、[4]を参照してください。
In this flow, notice that the Contact returned by NGW 1 in messages F7-11 is sip:ngw1@a.example.com. This is because NGW 1 only accepts SIP messages that come through Proxy 1 - any direct signaling will be ignored. Since this Contact URI may be used outside of this dialog and must be routable (Section 8.1.1.8 in RFC 3261 [2]) the Contact URI for NGW 1 must resolve to Proxy 1. This Contact URI resolves via DNS to Proxy 1 (sip:ss1.a.example.com) which then resolves it to sip:ngw1.a.example.com which is the address of NGW 1.
このフローでは、メッセージF7-11でNGW 1によって返された連絡先は、sip：ngw1@a.example.comであることに注意してください。これは、NGW 1がプロキシ1を介して来るSIPメッセージのみを受け入れるためです - 直接シグナルは無視されます。この連絡先URIはこのダイアログの外で使用され、ルーティング可能である必要があるため（RFC 3261 [2] in RFC 3261 [2]）、NGW 1の連絡先URIがプロキシに解決する必要があります。：ss1.a.example.com）は、sip：ngw1.a.example.comに解決します。これはNGW 1のアドレスです。
This flow shows TCP transport.
このフローは、TCP輸送を示しています。
Message Details
メッセージの詳細
F1 INVITE Alice -> Proxy 1
F1アリスを招待 - >プロキシ1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Proxy-Authorization: Digest username="alice", realm="a.example.com",
nonce="dc3a5ab25302aa931904ba7d88fa1cf5", opaque="",
uri="sip:+19725552222@ss1.a.example.com;user=phone",
response="ccdca50cb091d587421457305d097458c"
Content-Type: application/sdp
Content-Length: 154
v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=-
c=IN IP4 client.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F2 100 Trying Proxy 1 -> Alice
SIP/2.0 100 Trying
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
/* Proxy 1 uses a Location Service function to determine the gateway
for terminating this call. The call is forwarded to NGW 1. Client
for A prepares to receive data on port 49172 from the
network.*/
F3 INVITE Proxy 1 -> NGW 1
F3招待プロキシ1-> NGW 1
INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 100 Trying NGW 1 -> Proxy 1
f4 100 NGW 1->プロキシ1を試します
SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> Bob
f5 iam ngw 1-> bob
IAM
CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 ACM Bob -> NGW 1
F6 ACM BOB-> NGW 1
ACM
ACM
F7 183 Session Progress NGW 1 -> Proxy 1
F7 183セッションの進行NGW 1->プロキシ1
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* NGW 1 sends PSTN audio (ringing) in the RTP path to A */
F8 183 Session Progress Proxy 1 -> Alice
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F9 ANM Bob -> NGW 1
f9 anm bob-> ngw 1
ANM
ANM
F10 200 OK NGW 1 -> Proxy 1
F10 200 OK NGW 1->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 gw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = -c = In ip4 gw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F11 200 OK Proxy 1 -> Alice
F11 200 OKプロキシ1->アリス
SIP/2.0 200 OK
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F12 ACK Alice -> Proxy 1
F12 ACKアリス - >プロキシ1
ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
F13 ACK Proxy 1 -> NGW 1
F13 ACKプロキシ1-> NGW 1
ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
/* Alice Hangs Up with Bob. */
F14 BYE Alice -> Proxy 1
F14 Bye Alice-> Proxy 1
BYE sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F15 BYE Proxy 1 -> NGW 1
F15 Bye Proxy 1-> NGW 1
BYE sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F16 200 OK NGW 1 -> Proxy 1
F16 200 OK NGW 1->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F17 200 OK Proxy 1 -> A
F17 200 OKプロキシ1-> a
SIP/2.0 200 OK
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F18 REL NGW 1 -> B
f18 rel ngw 1-> b
REL CauseCode=16 Normal
rel causecode = 16正常
F19 RLC B -> NGW 1
F19 RLC B-> NGW 1
RLC
RLC
Alice Proxy 1 GW 1 PBX C
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| SETUP F5 |
| | |--------------->|
| | | CALL PROC F6 |
| | |<---------------|
| | | PROGress F7 |
| | 180 F8 |<---------------|
| 180 F9 |<---------------| |
|<---------------| | |
| | | One Way Voice |
| | |<===============|
| | | CONNect F10 |
| | |<---------------|
| | | CONNect ACK F11|
| | 200 F12 |--------------->|
| 200 F13 |<---------------| |
|<---------------| | |
| ACK F14 | | |
|--------------->| ACK F15 | |
| |--------------->| |
| Both Way RTP Media | Both Way Voice |
|<===============================>|<==============>|
| BYE F16 | | |
|--------------->| BYE F17 | |
| |--------------->| |
| | 200 F18 | |
| 200 F19 |<---------------| DISConnect F20 |
|<---------------| |--------------->|
| | | RELease F21 |
| | |<---------------|
| | | RELease COM F22|
| | |--------------->|
| | | |
Alice is a SIP device while Carol is connected via a Gateway (GW 1) to a PBX. The PBX connection is via a ISDN trunk group. Alice dials Carol's telephone number (918-555-3333) which is globalized and put into a SIP URI.
アリスはSIPデバイスであり、キャロルはゲートウェイ（GW 1）を介してPBXに接続されています。PBX接続は、ISDNトランクグループを介して行われます。アリスはキャロルの電話番号（918-555-3333）にダイヤルし、グローバル化され、SIP URIに入れられます。
The host portion of the Request-URI in the INVITE F3 is used to identify the context (customer, trunk group, or line) in which the private number 444-3333 is valid. Otherwise, this INVITE message could get forwarded by GW 1 and the context of the digits could become lost and the call unroutable.
招待F3のリクエスト-URIのホスト部分は、プライベート番号444-3333が有効なコンテキスト（顧客、トランクグループ、またはライン）を識別するために使用されます。それ以外の場合、この招待メッセージはGW 1によって転送される可能性があり、数字のコンテキストが失われ、コールが不十分になる可能性があります。
Proxy 1 looks up the telephone number and locates the gateway that serves Carol. Carol is identified by its extension (444-3333) in the Request-URI sent to GW 1.
Proxy 1は電話番号を調べ、キャロルにサービスを提供するゲートウェイを見つけます。キャロルは、GW 1に送信されたリクエスト-URIでその拡張（444-3333）によって識別されます。
Note that the Contact URI for GW 1, as used in messages F8, F9, F12, and F13, is sips:4443333@gw1.a.example.com, which resolves directly to the gateway.
メッセージF8、F9、F12、およびF13で使用されるGW 1の接触URIは、SIP：4443333@gw1.a.example.comであり、ゲートウェイに直接解決します。
This flow shows the use of Secure SIP (sips) URIs.
このフローは、安全なSIP（SIP）URIの使用を示しています。
Message Details
メッセージの詳細
F1 INVITE Alice -> Proxy 1
F1アリスを招待 - >プロキシ1
INVITE sips:+19185553333@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:alice@client.a.example.com>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="qo0dc3a5ab22aa931904badfa1cf5j9h",
opaque="", uri="sips:+19185553333@ss1.a.example.com;user=phone",
response="6c792f5c9fa360358b93c7fb826bf550"
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F2 100 Trying Proxy 1 -> Alice
f2 100プロキシ1->アリスを試す
SIP/2.0 100 Trying
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Content-Length: 0
F3 INVITE Proxy 1 -> GW 1
F3招待プロキシ1-> GW 1
INVITE sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:alice@client.a.example.com>
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 100 Trying GW -> Proxy 1
F4 100 GW->プロキシ1を試します
SIP/2.0 100 Trying
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Content-Length: 0
F5 SETUP GW 1 -> Carol
Protocol discriminator=Q.931 Message type=SETUP Bearer capability: Information transfer capability=0 (Speech) or 16 (3.1 kHz audio) Channel identification=Preferred or exclusive B-channel Progress indicator=1 (Call is not end-to-end ISDN;further call progress information may be available inband) Called party number: Type of number unknown Digits=444-3333
プロトコル差別= Q.931メッセージタイプ=セットアップベアラー機能：情報転送能力= 0（スピーチ）または16（3.1 kHzオーディオ）チャネル識別=優先または排他的なBチャンネルの進行状況インジケーター= 1（コールはエンドツーエンドではありませんISDN;さらなる通話進行状況情報はバンドで利用可能になる場合があります）パーティー番号：数字のタイプ不明な数字= 444-3333
F6 CALL PROCeeding Carol-> GW 1
f6コールプローチングキャロル - > GW 1
Protocol discriminator=Q.931 Message type=CALL PROC Channel identification=Exclusive B-channel
プロトコル識別子= Q.931メッセージタイプ=コールProcチャネル識別=排他的なBチャンネル
F7 PROGress Carol-> GW 1
F7 Progress Carol-> GW 1
Protocol discriminator=Q.931 Message type=PROG Progress indicator=1 (Call is not end-to-end ISDN;further call progress information may be available inband)
プロトコル識別子= Q.931メッセージタイプ= Prog Progress Indicator = 1（コールはエンドツーエンドISDNではありません。コールの進行状況情報はINBANDを利用できる場合があります）
F8 180 Ringing GW 1 -> Proxy 1
F8 180リンギングGW 1->プロキシ1
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com>
Content-Length: 0
F9 180 Ringing Proxy 1 -> Alice
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com>
Content-Length: 0
F10 CONNect Carol-> GW 1
F10 CONNECT CAROL-> GW 1
Protocol discriminator=Q.931 Message type=CONN
プロトコル識別子= Q.931メッセージタイプ= conn
F11 CONNect ACK GW 1 -> Carol
F11 Connect ACK GW 1-> Carol
Protocol discriminator=Q.931 Message type=CONN ACK
プロトコル差別= Q.931メッセージタイプ= conn ack
F12 200 OK GW 1 -> Proxy 1
F12 200 OK GW 1->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com>
Content-Type: application/sdp
Content-Length: 144
v=0
o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com
s=-
c=IN IP4 gw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F13 200 OK Proxy 1 -> Alice
F13 200 OKプロキシ1->アリス
SIP/2.0 200 OK
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sips:4443333@gw1.a.example.com>
Content-Type: application/sdp
Content-Length: 144
v=0 o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com s=- c=IN IP4 gw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 IN IP4 gw1.a.example.com s = -c = In ip4 gw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F14 ACK Alice -> Proxy 1
F14 ACKアリス - >プロキシ1
ACK sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 ACK
Content-Length: 0
F15 ACK Proxy 1 -> GW 1
ACK sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 ACK
Content-Length: 0
/* Alice Hangs Up with Bob. */
F16 BYE Alice -> Proxy 1
F16 Bye Alice-> Proxy 1
BYE sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 3 BYE
Content-Length: 0
F17 BYE Proxy 1 -> GW 1
F17 Bye Proxy 1-> GW 1
BYE sips:4443333@gw1.a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 3 BYE
Content-Length: 0
F18 200 OK GW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 3 BYE
Content-Length: 0
F19 200 OK Proxy 1 -> A
F19 200 OKプロキシ1-> a
SIP/2.0 200 OK
Via: SIP/2.0/TLS client.a.example.com:5061;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sips:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Carol <sips:+19185553333@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 3 BYE
Content-Length: 0
F20 DISConnect GW 1 -> Carol
F20 GW 1-> Carolを切断します
Protocol discriminator=Q.931 Message type=DISC Cause=16 (Normal clearing)
プロトコル識別子= Q.931メッセージタイプ=ディスク原因= 16（通常のクリアリング）
F21 RELease Carol-> GW 1
F21リリースキャロル - > GW 1
Protocol discriminator=Q.931 Message type=REL
プロトコル識別子= Q.931メッセージタイプ= rel
F22 RELease COMplete GW 1 -> Carol
F22リリース完全GW 1->キャロル
Protocol discriminator=Q.931 Message type=REL COM
プロトコル識別子= Q.931メッセージタイプ= rel com
Alice Proxy 1 NGW 1 NGW 2 Switch B
| | | | |
| INVITE F1 | | | |
|------------->| | | |
| | INVITE F2 | | |
| 100 F3 |------------->| | |
|<-------------| 503 F4 | | |
| |<-------------| | |
| | ACK F5 | | |
| |------------->| | |
| | INVITE F6 | |
| |---------------------------->| IAM F7 |
| | |------------->|
| | | ACM F8 |
| | 183 F9 |<-------------|
| 183 F10 |<----------------------------| |
|<-------------| | |
| Two Way RTP Media | One Way Voice|
|<==========================================>|<=============|
| | | ANM F11 |
| | 200 F12 |<-------------|
| 200 F13 |<----------------------------| |
|<-------------| | |
| ACK F14 | | |
|------------->| ACK F15 | |
| |---------------------------->| |
| Both Way RTP Media |Both Way Voice|
|<==========================================>|<============>|
| BYE F16 | | |
|------------->| BYE F17 | |
| |---------------------------->| |
| | 200 F18 | |
| 200 F19 |<----------------------------| REL F20 |
|<-------------| |------------->|
| | | RLC F21 |
| | |<-------------|
| | | |
Alice calls Bob through Proxy 1. Proxy 1 tries to route to a Network Gateway NGW 1. NGW 1 is not available and responds with a 503 Service Unavailable (F4). The call is then routed to Network Gateway NGW 2. Bob answers the call. The call is terminated when Alice disconnects the call. NGW 2 and Bob's telephone switch use ANSI ISUP signaling.
アリスはプロキシ1を通じてボブに電話をかけます。プロキシ1は、ネットワークゲートウェイNGWへのルーティングを試みます。NGW1は利用できず、503サービスが利用できない（F4）。その後、コールはネットワークゲートウェイNGW 2にルーティングされます。BOBはコールに応答します。アリスがコールを切断すると、コールが終了します。NGW 2とボブの電話スイッチは、ANSI ISUPシグナル伝達を使用します。
NGW 2 also only accepts SIP messages that come through Proxy 1, so the Contact URI sip:ngw2@a.example.com is used in this flow.
NGW 2はまた、プロキシ1を介して来るSIPメッセージのみを受け入れるため、このフローではuri sip：ngw2@a.example.comに連絡しています。
This flow shows UDP transport.
このフローは、UDP輸送を示しています。
Message Details
メッセージの詳細
F1 INVITE Alice -> Proxy 1
F1アリスを招待 - >プロキシ1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="b59311c3ba05b401cf80b2a2c5ac51b0",
opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone",
response="ba6ab44923fa2614b28e3e3957789ab0"
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine where B is
located. Proxy 1 receives a primary route NGW 1 and a secondary
route NGW 2. NGW 1 is tried first */
F2 INVITE Proxy 1 -> NGW 1
F2招待プロキシ1-> NGW 1
INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com>
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F3 100 Trying Proxy 1 -> Alice
F3 100プロキシ1->アリスを試してください
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F4 503 Service Unavailable NGW 1 -> Proxy 1
F4 503サービス利用不可NGW 1->プロキシ1
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 ACK Proxy 1 -> NGW 1
ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com>;user=phone>
;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
/* Proxy 1 now tries secondary route to NGW 2 */
F6 INVITE Proxy 1 -> NGW 2
F6招待プロキシ1-> NGW 2
INVITE sip:+19725552222@ngw2.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com>
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F7 IAM NGW 2 -> Bob
f7 iam ngw 2->ボブ
IAM
CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National
F8 ACM Bob -> NGW 2
ACM
ACM
F9 183 Session Progress NGW 2 -> Proxy 1
F9 183セッションの進捗NGW 2->プロキシ1
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw2@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com s=- c=IN IP4 ngw2.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw2.a.example.com s = - c = in ip4 ngw2.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* RTP packets are sent by GW to A for audio (e.g. ring tone) */
F10 183 Session Progress Proxy 1 -> Alice
F10 183セッション進行プロキシ1->アリス
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw2@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com s=- c=IN IP4 ngw2.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw2.a.example.com s = - c = in ip4 ngw2.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F11 ANM Bob -> NGW 2
F11 ANM BOB-> NGW 2
ANM
ANM
F12 200 OK NGW 2 -> Proxy 1
F12 200 OK NGW 2->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw2@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com s=- c=IN IP4 ngw2.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw2.a.example.com s = - c = in ip4 ngw2.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F13 200 OK Proxy 1 -> Alice
F13 200 OKプロキシ1->アリス
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw2@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw2.a.example.com s=- c=IN IP4 ngw2.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw2.a.example.com s = - c = in ip4 ngw2.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F14 ACK Alice -> Proxy 1
F14 ACKアリス - >プロキシ1
ACK sip:ngw2@a.example.com SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
F15 ACK Proxy 1 -> NGW 2
F15 ACKプロキシ1-> NGW 2
ACK sip:ngw2@a.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
/* RTP streams are established between A and B(via the GW) */
/* Alice Hangs Up with Bob. */
F16 BYE Alice -> Proxy 1
F16 Bye Alice-> Proxy 1
BYE sip:ngw2@a.example.com SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
Route: <ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F17 BYE Proxy 1 -> NGW 2
F17 Bye Proxy 1-> NGW 2
BYE sip:ngw2@a.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F18 200 OK NGW 2 -> Proxy 1
F18 200 OK NGW 2->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F19 200 OK Proxy 1 -> Alice
F19 200 OKプロキシ1->アリス
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 BYE
Content-Length: 0
F20 REL NGW 2 -> B
f20 rel ngw 2-> b
REL CauseCode=16 Normal
rel causecode = 16正常
F21 RLC B -> NGW 2
F21 RLC B-> NGW 2
RLC
RLC
Alice DNS Server Proxy 3 Bob
| | | |
| ENUM Query F1 | | |
|--------------->| | |
| Response F2 | | |
|<---------------| | |
| INVITE F3 | |
|-------------------------------->| INVITE F4 |
| 100 F5 |--------------->|
|<--------------------------------| 180 F6 |
| 180 F7 |<---------------|
|<--------------------------------| |
| | 200 F8 |
| 200 F9 |<---------------|
|<--------------------------------| |
| ACK F10 | |
|-------------------------------->| ACK F11 |
| |--------------->|
| Both Way RTP Media |
|<================================================>|
| | BYE F12 |
| BYE F13 |<---------------|
|<--------------------------------| |
| 200 F14 | |
|-------------------------------->| 200 F15 |
| |--------------->|
| | |
In this scenario, Alice places a call to Bob by dialing Bob's telephone number (9725552222). Alice's UA converts the phone number to an E.164 number (+19725552222), and performs an ENUM query [9] on the E.164 number (2.2.2.2.5.5.5.2.7.9.1.e164.arpa), which returns a NAPTR record containing a SIP AOR URI for Bob (sip:+19725552222@b.example.com). As a result, Alice's UA sends an INVITE and the call completes over IP bypassing the PSTN.
このシナリオでは、アリスはボブの電話番号（9725552222）をダイヤルしてボブに電話をかけます。Alice's UAは電話番号をE.164番号（19725552222）に変換し、E.164番号（2.2.2.2.5.5.5.5.55.2.7.9.1.e164.e164.ARPA）で列挙クエリ[9]を実行します。BOBのSIP AOR URIを含むNAPTRレコード（SIP：19725552222@b.example.com）。その結果、アリスのUAは招待状を送信し、PSTNをバイパスするIPでコールが完了します。
The call is terminated when Bob sends a BYE message.
ボブがさようならメッセージを送信すると、呼び出しが終了します。
Message Details
メッセージの詳細
F1 ENUM Query Alice -> DNS Server
f1 enum query alice-> dnsサーバー
2.2.2.2.5.5.5.2.7.9.1.e164.arpa F2 ENUM NAPTR Set DNS Server -> Alice
2.2.2.2.5.5.5.5.5.2.7.9.1.E164.ARPA F2 ENUM NAPTR SET DNS SERVER-> ALICE
$ORIGIN 2.2.2.2.5.5.5.2.7.9.1.e164.arpa. IN NAPTR 100 10 "u" "sip+E2U" "!^.*$!sip:+19725552222@b.example.com!".
$ origin 2.2.2.2.5.5.5.5.2.7.9.1.e164.arpa。in naptr 100 10 "u" "sip e2u" "！^。*$！sip：197255222@b.example.com！"。
F3 INVITE Alice -> Proxy 3
F3アリスを招待 - >プロキシ3
INVITE sip:+19725552222@b.example.com SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <tel:+19725552222>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sip:+13145551111@client.a.example.com>
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 INVITE Proxy 3 -> Bob
F4招待プロキシ3->ボブ
INVITE sip:+19725552222@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <tel:+19725552222>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sip:+13145551111@client.a.example.com>
Content-Type: application/sdp
Content-Length: 154
v=0
o=UserA 2890844526 2890844526 IN IP4 client.a.example.com
s=-
c=IN IP4 client.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F5 100 Trying Proxy 3 -> Alice
F5 100プロキシ3->アリス
SIP/2.0 100 Trying
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <tel:+19725552222>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Content-Length: 0
F6 180 Ringing B -> Proxy 3
F6 180リンギングB->プロキシ3
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.233
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sip:+19725552222@client.b.example.com>
Content-Length: 0
F7 180 Ringing Proxy 3 -> Alice
F7 180リンギングプロキシ3->アリス
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sip:+19725552222@client.b.example.com>
Content-Length: 0
F8 200 OK Bob -> Proxy 3
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.233
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sip:+19725552222@client.b.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 151
v=0 o=bob 2890844527 2890844527 IN IP4 client.b.example.com s=- c=IN IP4 client.b.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = BOB 2890844527 2890844527 IN IP4 client.b.example.com s = -c = In ip4 client.b.example.com t = 0 0 M = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F9 200 OK Proxy -> Alice
F9 200 OKプロキシ - >アリス
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 INVITE
Contact: <sip:+19725552222@client.b.example.com>
Content-Type: application/sdp
Content-Length: 151
v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=-
c=IN IP4 192.0.2.100
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F10 ACK Alice -> Proxy 3
ACK sip:+19725552222@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9
Max-Forwards: 70
Route: <sip:ss3.b.example.com;lr>
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 ACK
Content-Length: 0
F11 ACK Proxy 3 -> Bob
F11 ACKプロキシ3->ボブ
ACK sip:+19725552222@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bq9
;received=192.0.2.101
Max-Forwards: 69
From: <sip:+13145551111@a.example.com>;tag=9fxced76sl
To: <tel:+19725552222>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 2 ACK
Content-Type: application/sdp
Content-Length: 0
/* RTP streams are established between A and B*/
/* User B Hangs Up with User A. */
F12 BYE Bob -> Proxy 3
f12 bye bob-> proxy 3
BYE sip:+13145551111@client.a.example.com SIP/2.0
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
Max-Forwards: 70
Route: <sip:ss3.b.example.com;lr>
From: <tel:+19725552222>;tag=314159
To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 BYE
Content-Length: 0
F13 BYE Proxy 3 -> Alice
F13 Bye Proxy 3-> Alice
BYE sip:+13145551111@client.a.example.com SIP/2.0
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.100
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
Max-Forwards: 69
From: <tel:+19725552222>;tag=314159
To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 BYE
Content-Length: 0
F14 200 OK Alice -> Proxy 3
F14 200 OKアリス - >プロキシ3
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss3.b.example.com:5060;branch=z9hG4bK721e418c4.1
;received=192.0.2.233
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
;received=192.0.2.100
From: <tel:+19725552222>;tag=314159
To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 BYE
Content-Length: 0
F15 200 OK Proxy 3 -> Bob
F15 200 OKプロキシ3->ボブ
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.b.example.com:5060;branch=z9hG4bKfgaw2
;received=192.0.2.100
From: <tel:+19725552222>;tag=314159
To: <sip:+13145551111@a.example.com>;tag=9fxced76sl
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 BYE
Content-Length: 0
Alice Proxy 1 NGW 1 Bob
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | ACM F6 |
| | 183 F7 |<---------------|
| 183 F8 |<---------------| |
|<---------------| | |
| Two Way RTP Media | One Way Voice |
|<===============================>|<===============|
| Treatment Applied |
|<=================================================|
| CANCEL F9 | | |
|--------------->| | |
| 200 F10 | | |
|<---------------| CANCEL F11 | |
| |--------------->| |
| | 200 F12 | |
| |<---------------| REL F13 |
| | |--------------->|
| | | RLC F14 |
| | 487 F15 |<---------------|
| |<---------------| |
| | ACK F16 | |
| 487 F17 |--------------->| |
|<---------------| | |
| ACK F18 | | |
|--------------->| | |
| | | |
Alice calls Bob in the PSTN through a proxy server Proxy 1 and a Network Gateway NGW 1. The call is rejected by the PSTN with an in-band treatment (tone or recording) played. Alice hears the treatment and then hangs up, which results in a CANCEL (F9) being sent to terminate the call. (A BYE is not sent since no final response was ever received by Alice.) Message Details
アリスは、プロキシサーバープロキシ1およびネットワークゲートウェイNGW 1を介してPSTNのボブに電話をかけます。アリスは治療を聞いてから電話を切るため、コールを終了するためにキャンセル（F9）が送信されます。（アリスが最終的な応答を受け取ったことがないため、さようならは送信されません。）メッセージの詳細
F1 INVITE Alice -> Proxy 1
F1アリスを招待 - >プロキシ1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="01cf8311c3b0b2a2c5ac51bb59a05b40",
opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone",
response="e178fbe430e6680a1690261af8831f40"
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F2 100 Trying Proxy 1 -> A
F2 100プロキシ1-> a
SIP/2.0 100 Trying
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to NGW
1. Client for A prepares to receive data on port 49172 from the
network. */
F3 INVITE Proxy 1 -> NGW 1
INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com>
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 100 Trying NGW 1 -> Proxy 1
f4 100 NGW 1->プロキシ1を試します
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> Bob
f5 iam ngw 1-> bob
IAM
CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 ACM Bob -> NGW 1
ACM
ACM
F7 183 Session Progress NGW 1 -> Proxy 1
F7 183セッションの進行NGW 1->プロキシ1
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F8 183 Session Progress Proxy 1 -> Alice
F8 183セッション進行プロキシ1->アリス
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* Caller hears the recorded announcement, then hangs up */
F9 CANCEL Alice -> Proxy 1
f9アリスキャンセル - >プロキシ1
CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 CANCEL
Content-Length: 0
F10 200 OK Proxy 1 -> A
F10 200 OKプロキシ1-> a
SIP/2.0 200 OK
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 CANCEL
Content-Length: 0
F11 CANCEL Proxy 1 -> NGW 1
F11キャンセルプロキシ1-> NGW 1
CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 CANCEL
Content-Length: 0
F12 200 OK NGW 1 -> Proxy 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 CANCEL
Content-Length: 0
F13 REL NGW 1 -> B
f13 rel ngw 1-> b
REL CauseCode=18 No user responding
rel causecode = 18ユーザー応答なし
F14 RLC B -> NGW 1
F14 RLC B-> NGW 1
RLC
RLC
F15 487 Request Terminated NGW 1 -> Proxy 1
F15 487リクエスト終了NGW 1->プロキシ1
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F16 ACK Proxy 1 -> NGW 1
F16 ACKプロキシ1-> NGW 1
ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
F17 487 Request Terminated Proxy 1 -> A
F17 487リクエスト終了プロキシ1-> a
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F18 ACK Alice -> Proxy 1
F18 ACKアリス - >プロキシ1
ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
Alice Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | REL(1) F6 |
| | |<---------------|
| | | RLC F7 |
| | 404 F8 |--------------->|
| |<---------------| |
| | ACK F9 | |
| |--------------->| |
| 404 F10 | | |
|<---------------| | |
| ACK F11 | | |
|--------------->| | |
| | | |
Alice calls PSTN Bob through a Proxy Server Proxy 1 and a Network Gateway NGW 1. The call is rejected by the PSTN with a ANSI ISUP Release message REL containing a specific Cause code. This cause value (1) is mapped by the Gateway to a SIP 404 Address Incomplete response which is proxied back to Alice. For more details of ISUP cause value to SIP response mapping, refer to [4].
アリスは、プロキシサーバープロキシ1およびネットワークゲートウェイNGWを介してPSTN BOBを呼び出します。PSTNによって、特定の原因コードを含むANSI ISUPリリースメッセージを使用してPSTNによって拒否されます。この原因値（1）は、AliceにプロキシされたSIP 404アドレスの不完全な応答へのゲートウェイによってマッピングされます。ISUPの詳細については、SIP応答マッピングの値を原因で、[4]を参照してください。
Message Details
メッセージの詳細
F1 INVITE Alice -> Proxy 1
F1アリスを招待 - >プロキシ1
INVITE sip:+44-1234@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="j1c3b0b01cf832da2c5ac51bb59a05b40",
opaque="", uri="sip:+44-1234@ss1.a.example.com;user=phone",
response="a451358d46b55512863efe1dccaa2f42"
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F2 100 Trying Proxy 1 -> A
F2 100プロキシ1-> a
SIP/2.0 100 Trying
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to NGW1.
Client for A prepares to receive data on port 49172 from the network.
*/
F3 INVITE Proxy 1 -> NGW 1
F3招待プロキシ1-> NGW 1
INVITE sip:+44-1234@ngw1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 154
v=0
o=alice 2890844526 2890844526 IN IP4 client.a.example.com
s=-
c=IN IP4 client.a.example.com
t=0 0
m=audio 49172 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying NGW 1 -> Proxy 1
f4 100 NGW 1->プロキシ1を試します
SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> Bob
f5 iam ngw 1-> bob
IAM
CdPN=44-1234,NPI=E.164,NOA=International
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 REL Bob -> NGW 1
f6 rel bob-> ngw 1
REL CauseValue=1 Unallocated number
Rel Causevalue = 1未割り当て番号
F7 RLC NGW 1 -> Bob
F7 RLC NGW 1->ボブ
RLC
RLC
/* Network Gateway maps CauseValue=1 to the SIP message 404 Not
Found */
F8 404 Not Found NGW 1 -> Proxy 1
f8 404 not not not ngw 1-> proxy1
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Error-Info: <sip:not-found-ann@ann.a.example.com>
Content-Length: 0
F9 ACK Proxy 1 -> NGW 1
F9 ACKプロキシ1-> NGW 1
ACK sip:+44-1234@ngw1.a.example.com;user=phone SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
F10 404 Not Found Proxy 1 -> Alice
F10 404プロキシ1->アリス
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Error-Info: <sip:not-found-ann@ann.a.example.com>
Content-Length: 0
F11 ACK Alice -> Proxy 1
F11 ACKアリス - >プロキシ1
ACK sip:+44-1234@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+44-1234@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
Alice Proxy 1 NGW 1 Switch B
| | | |
| INVITE F1 | | |
|--------------->| | |
| 100 F2 | | |
|<---------------| INVITE F3 | |
| |--------------->| |
| | 100 F4 | |
| |<---------------| IAM F5 |
| | |--------------->|
| | | ACM F6 |
| | 183 F7 |<---------------|
| 183 F8 |<---------------| |
|<---------------| | |
| | Timer on NGW 1 Expires |
| | | |
| | | REL F9 |
| | |--------------->|
| | | RLC F10 |
| | 480 F11 |<---------------|
| |<---------------| |
| | ACK F12 | |
| |--------------->| |
| 480 F13 | | |
|<---------------| | |
| ACK F14 | | |
|--------------->| | |
Alice calls Bob in the PSTN through a proxy server Proxy 1 and Network Gateway NGW 1. The call is released by the Gateway after a timer expires due to no ANswer Message (ANM) being received. The Gateway sends an ISUP Release REL message to the PSTN and a 480 Temporarily Unavailable response to Alice in the SIP network.
アリスは、プロキシサーバープロキシ1およびネットワークゲートウェイNGW 1を介してPSTNのボブを呼び出します。回答メッセージ（ANM）が受信されないためにタイマーが期限切れになった後、コールはゲートウェイによってリリースされます。ゲートウェイは、PSTNにISUPリリースのRelメッセージを送信し、SIPネットワークでAliceに一時的に利用できない480を一時的に利用できません。
Message Details
メッセージの詳細
F1 INVITE Alice -> Proxy 1
F1アリスを招待 - >プロキシ1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Proxy-Authorization: Digest username="alice",
realm="a.example.com", nonce="da2c5ac51bb59a05j1c3b0b01cf832b40",
opaque="", uri="sip:+19725552222@ss1.a.example.com;user=phone",
response="579cb9db184cdc25bf816f37cbc03c7d"
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to NGW
1. Client for A prepares to receive data on port 49172 from the
network.*/
F2 100 Trying Proxy 1 -> A
F2 100プロキシ1-> a
SIP/2.0 100 Trying
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F3 INVITE Proxy 1 -> NGW 1
INVITE sip:+19725552222@ngw1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:alice@client.a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 154
v=0 o=alice 2890844526 2890844526 IN IP4 client.a.example.com s=- c=IN IP4 client.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = Alice 2890844526 2890844526 in ip4 client.a.example.com s = - c = in ip4 client.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 100 Trying NGW 1 -> Proxy 1
f4 100 NGW 1->プロキシ1を試します
SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 IAM NGW 1 -> Bob
f5 iam ngw 1-> bob
IAM
CdPN=972-555-2222,NPI=E.164,NOA=National
CgPN=314-555-1111,NPI=E.164,NOA=National
F6 ACM Bob -> NGW 1
ACM
ACM
F7 183 Session Progress NGW 1 -> Proxy 1
F7 183セッションの進行NGW 1->プロキシ1
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F8 183 Session Progress Proxy 1 -> Alice
F8 183セッション進行プロキシ1->アリス
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
Record-Route: <sip:ss1.a.example.com;lr>
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
/* After NGW 1's timer expires, Network Gateway sends REL to ISUP
network and 480 to SIP network */
F9 REL NGW 1 -> Bob
f9 rel ngw 1-> bob
REL
rel
CauseCode=18 No user responding
CauseCode = 18ユーザー応答なし
F10 RLC Bob -> NGW 1
F10 RLC BOB-> NGW 1
RLC
RLC
F11 480 Temporarily Unavailable NGW 1 -> Proxy 1
F11 480一時的に利用できないNGW 1->プロキシ1
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Error-Info: <sip:temp-unavail-ann@ann.a.example.com>
Content-Length: 0
F12 ACK Proxy 1 -> NGW 1
F12 ACKプロキシ1-> NGW 1
ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
F13 480 Temporarily Unavailable F13 Proxy 1 -> Alice
F13 480一時的に利用できないF13プロキシ1->アリス
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
;received=192.0.2.101
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 INVITE
Error-Info: <sip:temp-unavail-ann@ann.a.example.com>
Content-Length: 0
F14 ACK Alice -> Proxy 1
F14 ACKアリス - >プロキシ1
ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/TCP client.a.example.com:5060;branch=z9hG4bK74bf9
From: Alice <sip:+13145551111@ss1.a.example.com;user=phone>
;tag=9fxced76sl
To: Bob <sip:+19725552222@ss1.a.example.com;user=phone>
;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@a.example.com
CSeq: 1 ACK
Content-Length: 0
In these scenarios, Alice is placing calls from the PSTN to Bob in a SIP network. Alice's telephone switch signals to a Network Gateway (NGW 1) using ANSI ISUP.
これらのシナリオでは、アリスはSIPネットワークでPSTNからBOBに電話をかけています。ANSI ISUPを使用して、アリスの電話スイッチはネットワークゲートウェイ（NGW 1）に合図します。
Since the called SIP User Agent does not send in-band signaling information, no early media path needs to be established on the IP side. As a result, the 183 Session Progress response is not used. However, NGW 1 will establish a one way speech path prior to call completion, and generate ringing for the PSTN caller. Any tones or recordings are generated by NGW 1 and played in this speech path. When the call completes successfully, NGW 1 bridges the PSTN speech path with the IP media path.
呼び出されたSIPユーザーエージェントは帯域内のシグナリング情報を送信しないため、IP側に初期のメディアパスを確立する必要はありません。その結果、183セッションの進行状況応答は使用されません。ただし、NGW 1は、呼び出し完了の前に一方向の音声パスを確立し、PSTN発信者のリンギングを生成します。トーンまたは録音はNGW 1によって生成され、このスピーチパスで再生されます。コールが正常に完了すると、NGW 1はIPメディアパスでPSTNの音声パスをブリッジします。
To reduce the number of messages, only a single proxy server is shown in these flows, which means that the a.example.com proxy server has access to the b.example.com location service.
メッセージの数を減らすために、これらのフローに単一のプロキシサーバーのみが表示されます。つまり、a.example.comプロキシサーバーはb.example.comロケーションサービスにアクセスできます。
Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 180 F5 |
| | 180 F6 |<---------------|
| ACM F7 |<---------------| |
|<---------------| | |
| One Way Voice | | |
|<===============| | |
| Ringing Tone | | 200 F8 |
|<===============| 200 F9 |<---------------|
| |<---------------| |
| | ACK F10 | |
| ANM F12 |--------------->| ACK F11 |
|<---------------| |--------------->|
| Both Way Voice | Both Way RTP Media |
|<==============>|<===============================>|
| REL F13 | | |
|--------------->| | |
| RLC F14 | | |
|<---------------| BYE F15 | |
| |--------------->| BYE F16 |
| | |--------------->|
| | | 200 F17 |
| | 200 F18 |<---------------|
| |<---------------| |
| | | |
In this scenario, Alice from the PSTN calls Bob through a Network Gateway NGW1 and Proxy Server Proxy 1. When Bob answers the call, the media path is setup end-to-end. The call terminates when Alice hangs up the call, with Alice's telephone switch sending an ISUP RELease message that is mapped to a BYE by NGW 1.
このシナリオでは、PSTNのアリスは、ネットワークゲートウェイNGW1とプロキシサーバープロキシ1を介してボブを呼び出します。BOBが通話に応答すると、メディアパスはエンドツーエンドをセットアップします。アリスがコールを切るときにコールが終了し、アリスの電話スイッチがNGW 1によってさようならにマッピングされるISUPリリースメッセージを送信します。
Message Details
メッセージの詳細
F1 IAM Alice -> NGW 1
f1 iam alice-> ngw 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE Alice -> Proxy 1
F2アリスを招待 - >プロキシ1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to NGW
1. NGW 1 prepares to receive data on port 3456 from Alice.*/
F3 INVITE Proxy 1 -> Bob
INVITE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 100 Trying Bob -> Proxy 1
F4 100 BOBを試す - >プロキシ1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 180 Ringing Bob -> Proxy 1
F5 180リンギングボブ - >プロキシ1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com>
Content-Length: 0
F6 180 Ringing Proxy 1 -> NGW 1
F6 180リンギングプロキシ1-> NGW 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com>
Content-Length: 0
F7 ACM NGW 1 -> Alice
F7 ACM NGW 1->アリス
ACM
ACM
F8 200 OK Bob -> Proxy 1
F8 200 OK BOB->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
Contact: <sip:bob@client.b.example.com>
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 151
v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=-
c=IN IP4 client.b.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F9 200 OK Proxy 1 -> NGW 1
F9 200 OKプロキシ1-> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com>
Content-Type: application/sdp
Content-Length: 151
v=0 o=bob 2890844527 2890844527 IN IP4 client.b.example.com s=- c=IN IP4 client.b.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = BOB 2890844527 2890844527 IN IP4 client.b.example.com s = -c = In ip4 client.b.example.com t = 0 0 M = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F10 ACK NGW 1 -> Proxy 1
F10 ACK NGW 1->プロキシ1
ACK sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F11 ACK Proxy 1 -> Bob
F11 ACKプロキシ1->ボブ
ACK sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F12 ANM Bob -> NGW 1
F12 ANM BOB-> NGW 1
ANM
ANM
/* RTP streams are established between A and B (via the GW) */
/* Alice Hangs Up with Bob. */
F13 REL Alice -> NGW 1
F13 Rel Alice-> ngw 1
REL CauseCode=16 Normal
rel causecode = 16正常
F14 RLC NGW 1 -> Alice
F14 RLC NGW 1->アリス
RLC
RLC
F15 BYE NGW 1-> Proxy 1
f15 bye ngw 1->プロキシ1
BYE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
F16 BYE Proxy 1 -> Bob
F16 Bye Proxy 1-> Bob
BYE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
F17 200 OK Bob -> Proxy 1
F17 200 OK BOB->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
F18 200 OK Proxy 1 -> NGW 1
F18 200 OKプロキシ1-> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 200 F5 |
| | 200 F6 |<---------------|
| |<---------------| |
| | ACK F7 | |
| ANM F9 |--------------->| ACK F8 |
|<---------------| |--------------->|
| Both Way Voice | Both Way RTP Media |
|<==============>|<===============================>|
| REL F10 | | |
|--------------->| | |
| RLC F11 | | |
|<---------------| BYE F12 | |
| |--------------->| BYE F13 |
| | |--------------->|
| | | 200 F14 |
| | 200 F15 |<---------------|
| |<---------------| |
| | | |
This "fast answer" scenario is similar to 3.1., except that Bob immediately accepts the call, sending a 200 OK (F5) without sending a 180 Ringing response. The Gateway then sends an Answer Message (ANM) without sending an Address Complete Message (ACM). Note that for ETSI and some other ISUP variants, a CONnect message (CON) would be sent instead of the ANM.
この「高速回答」シナリオは3.1に似ています。ただし、ボブはすぐにコールを受け入れ、180のリンギング応答を送信せずに200 OK（F5）を送信します。次に、ゲートウェイは、アドレス完全なメッセージ（ACM）を送信せずに回答メッセージ（ANM）を送信します。ETSIおよびその他のISUPバリアントの場合、ANMの代わりに接続メッセージ（CON）が送信されることに注意してください。
Message Details
メッセージの詳細
F1 IAM Alice -> NGW 1
f1 iam alice-> ngw 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE NGW 1 -> Proxy 1
F2 NGW 1->プロキシ1を招待します
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine where B is
located. Based upon location analysis the call is forwarded to User
B. Bob prepares to receive data on port 3456 from Alice.*/
F3 INVITE Proxy 1 -> Bob
F3招待プロキシ1->ボブ
INVITE bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying Proxy 1 -> NGW 1
SIP/2.0 100 Trying
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 200 OK Bob -> Proxy 1
F5 200 OK BOB->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 151
v=0 o=bob 2890844527 2890844527 IN IP4 client.b.example.com s=- c=IN IP4 client.b.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = BOB 2890844527 2890844527 IN IP4 client.b.example.com s = -c = In ip4 client.b.example.com t = 0 0 M = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F6 200 OK Proxy 1 -> NGW 1
F6 200 OKプロキシ1-> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 151
v=0 o=bob 2890844527 2890844527 IN IP4 client.b.example.com s=- c=IN IP4 client.b.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = BOB 2890844527 2890844527 IN IP4 client.b.example.com s = -c = In ip4 client.b.example.com t = 0 0 M = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F7 ACK NGW 1 -> Proxy 1
F7 ACK NGW 1->プロキシ1
ACK bob@client.b.example.com SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F8 ACK Proxy 1 -> Bob
F8 ACKプロキシ1->ボブ
ACK bob@client.b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=130.131.132.14
Max-Forwards: 69
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F9 ANM Bob -> NGW 1
f9 anm bob-> ngw 1
ANM
ANM
/* RTP streams are established between A and B (via the GW) */
/* Alice Hangs Up with Bob. */
F10 REL ser Alice -> NGW 1
REL CauseCode=16 Normal
rel causecode = 16正常
F11 RLC NGW 1 -> Alice
F11 RLC NGW 1->アリス
RLC
RLC
F12 BYE NGW 1 -> Proxy 1
f12 bye ngw 1->プロキシ1
BYE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
F13 BYE Proxy 1 -> Bob
F13 Bye Proxy 1-> Bob
BYE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
F14 200 OK Bob -> Proxy 1
F14 200 OK BOB->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
F15 200 OK Proxy 1 -> NGW 1
F15 200 OKプロキシ1-> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
PBX A GW 1 Proxy 1 Bob
| | | |
| Seizure | | |
|--------------->| | |
| Wink | | |
|<---------------| | |
| MF Digits F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 180 F5 |
| | 180 F6 |<---------------|
| |<---------------| |
| One Way Voice | | |
|<===============| | |
| Ringing Tone | | 200 F7 |
|<===============| 200 F8 |<---------------|
| |<---------------| |
| | ACK F9 | |
| Seizure |--------------->| ACK F10 |
|<---------------| |--------------->|
| Both Way Voice | Both Way RTP Media |
|<==============>|<===============================>|
| Seizure Removal| | |
|--------------->| | |
| Seizure Removal| | |
|<---------------| BYE F11 | |
| |--------------->| BYE F12 |
| | |--------------->|
| | | 200 F13 |
| | 200 F14 |<---------------|
| |<---------------| |
| | | |
In this scenario, Alice dials from PBX A to Bob through GW 1 and Proxy 1. This is an example of a call that appears destined for the PSTN but is instead routed to a SIP Client.
このシナリオでは、アリスはPBX AからGW 1とプロキシ1を介してボブにダイヤルします。これは、PSTN向けに見えるが、代わりにSIPクライアントにルーティングされるコールの例です。
Signaling between PBX A and GW 1 is Feature Group B (FGB) circuit associated signaling, in-band Mult-Frequency (MF) outpulsing. After the receipt of the 180 Ringing from Bob, GW 1 generates a ringing tone for Alice.
PBX AとGW 1の間のシグナル伝達は、特徴グループB（FGB）回路関連シグナル伝達、インバンド多周頻度（MF）の補償です。ボブから180リンギングを受け取った後、GW 1はアリスのリンギングトーンを生成します。
Bob answers the call by sending a 200 OK. The call terminates when Alice hangs up, causing GW1 to send a BYE.
ボブは200 OKを送信して電話に応答します。コールは、アリスが電話を切ると終了し、GW1にさようならを送信します。
The Gateway can only identify the trunk group that the call came in on; it cannot identify the individual line on PBX A that is placing the call. The SIP URI used to identify the caller is shown in these flows as sip:551313@gw1.a.example.com.
ゲートウェイは、コールが入ったトランクグループのみを識別できます。コールを配置しているPBX Aの個々のラインを識別することはできません。発信者を識別するために使用されるSIP URIは、これらのフローにSIP：551313@gw1.a.example.comとして表示されます。
Message Details
メッセージの詳細
PBX Alice -> GW 1
PBXアリス - > GW 1
Seizure
発作
GW 1 -> PBX A
GW 1-> PBX a
Wink
ウィンク
F1 MF Digits PBX Alice -> GW 1
f1 mf digits pbx alice-> gw 1
KP 1 972 555 2222 ST
KP 1 972 555 2222 ST
F2 INVITE GW 1 -> Proxy 1
F2 GW 1->プロキシ1を招待します
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:551313@gw1.a.example.com;user=phone>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 gw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = -c = In ip4 gw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine where the
phone number +19725552222 is located. Based upon location
analysis the call is forwarded to SIP Bob. */
F3 INVITE Proxy 1 -> Bob
INVITE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:551313@gw1.a.example.com;user=phone>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com s=- c=IN IP4 gw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 IN IP4 gw1.a.example.com s = -c = In ip4 gw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 100 Trying Proxy 1 -> GW 1
F4 100プロキシ1-> GW 1を試します
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 180 Ringing Bob -> Proxy 1
F5 180リンギングボブ - >プロキシ1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com>
Content-Length: 0
F6 180 Ringing Proxy 1 -> GW 1
F6 180リンギングプロキシ1-> GW 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com>
Content-Length: 0
/* One way Voice path is established between GW and the PBX for
ringing. */
F7 200 OK Bob -> Proxy 1
F7 200 OK BOB->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
Contact: <sip:bob@client.b.example.com>
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 151
v=0
o=bob 2890844527 2890844527 IN IP4 client.b.example.com
s=-
c=IN IP4 client.b.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F8 200 OK Proxy 1 -> GW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com>
Content-Type: application/sdp
Content-Length: 151
v=0 o=bob 2890844527 2890844527 IN IP4 client.b.example.com s=- c=IN IP4 client.b.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = BOB 2890844527 2890844527 IN IP4 client.b.example.com s = -c = In ip4 client.b.example.com t = 0 0 M = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F9 ACK GW 1 -> Proxy 1
F9 ACK GW 1->プロキシ1
ACK sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F10 ACK Proxy 1 -> Bob
F10 ACKプロキシ1->ボブ
ACK sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
/* RTP streams are established between A and B (via the GW) */
/* Alice Hangs Up with Bob. */
F11 BYE GW 1 -> Proxy 1
F11バイGW 1->プロキシ1
BYE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
F12 BYE Proxy 1 -> Bob
F12バイプロキシ1->ボブ
BYE sip:bob@client.b.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
F13 200 OK Bob -> Proxy 1
F13 200 OK BOB->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
F14 200 OK Proxy 1 -> GW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
From: <sip:551313@gw1.a.example.com;user=phone>;tag=jwdkallkzm
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 2 BYE
Content-Length: 0
Switch A GW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| |
| | 604 F3 | |
| |<---------------| |
| | ACK F4 | |
| |--------------->| |
| REL F5 | | |
|<---------------| | |
| RLC F6 | | |
|--------------->| | |
| | | |
Alice attempts to place a call through Gateway GW 1 and Proxy 1, which is unable to find any routing for the number. The call is rejected by Proxy 1 with a REL message containing a specific Cause value mapped by the gateway based on the SIP error.
アリスは、Gateway GW 1とProxy 1を介して電話をかけようとします。これは、数のルーティングを見つけることができません。この呼び出しは、SIPエラーに基づいてゲートウェイによってマッピングされた特定の原因値を含むRELメッセージを使用して、Proxy 1によって拒否されます。
Message Details
メッセージの詳細
F1 IAM Alice -> GW 1
F1 Iam Alice-> GW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-9999,NPI=E.164,NOA=National
F2 INVITE Alice -> Proxy 1
F2アリスを招待 - >プロキシ1
INVITE sip:+1972559999@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s
To: <sip:+1972559999@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE
Contact:
<sip:+13145551111@gw1.a.example.com;user=phone;transport=tcp>
Content-Type: application/sdp
Content-Length: 144
v=0 o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com s=- c=IN IP4 gw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 IN IP4 gw1.a.example.com s = -c = In ip4 gw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service to find a route to +1-972-555-
9999. A route is not found, so Proxy 1 rejects the call. */
F3 604 Does Not Exist Anywhere Proxy 1 -> GW 1
F3 604はどこにも存在しませんプロキシ1-> GW 1
SIP/2.0 604 Does Not Exist Anywhere
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s
To: <sip:+1972559999@ss1.a.example.com;user=phone>;tag=6a34d410
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 INVITE
Error-Info: <sip:does-not-exist@ann.a.example.com>
Content-Length: 0
F4 ACK GW 1 -> Proxy 1
F4 ACK GW 1->プロキシ1
ACK sip:+1972559999@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@gw1.a.example.com;user=phone>;tag=076342s
To: <sip:+1972559999@ss1.a.example.com;user=phone>;tag=6a34d410
Call-ID: 4Fde34wkd11wsGFDs3@gw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F5 REL GW 1 -> Alice
REL CauseCode=1
rel causecode = 1
F6 RLC Alice -> GW 1
F6 RLCアリス - > GW 1
RLC
RLC
Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 600 F5 |
| | |<---------------|
| | | ACK F6 |
| | 600 F7 |--------------->|
| |<---------------| |
| | ACK F8 | |
| |--------------->| |
| REL(17) F9 | | |
|<---------------| | |
| RLC F10 | | |
|<-------------->| | |
| | | |
In this scenario, Alice calls Bob through Network Gateway NGW 1 and Proxy 1. The call is routed to Bob by Proxy 1. The call is rejected by Bob who sends a 600 Busy Everywhere response. The Gateway sends a REL message containing a specific Cause value mapped by the gateway based on the SIP error.
このシナリオでは、アリスはネットワークゲートウェイNGW 1とプロキシ1を介してボブを呼び出します。コールはプロキシ1によってボブにルーティングされます。ゲートウェイは、SIPエラーに基づいてゲートウェイによってマッピングされた特定の原因値を含むRELメッセージを送信します。
Since no interworking is indicated in the IAM (F1), the busy tone is generated locally by Alice's telephone switch. In some scenarios, the busy signal is generated by the Gateway since interworking is indicated. For more discussion on interworking, refer to [4].
IAM（F1）にはインターワーキングが示されていないため、ビジートーンはアリスの電話スイッチによってローカルに生成されます。一部のシナリオでは、インターワーキングが示されているため、忙しい信号がゲートウェイによって生成されます。インターワーキングの詳細については、[4]を参照してください。
Message Details
メッセージの詳細
F1 IAM Alice -> NGW 1
f1 iam alice-> ngw 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE Alice -> Proxy 1
F2アリスを招待 - >プロキシ1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 144
v=0 o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com s=- c=IN IP4 gw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 IN IP4 gw1.a.example.com s = -c = In ip4 gw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */
F3 INVITE F3 Proxy 1 -> Bob
F3招待F3プロキシ1->ボブ
INVITE bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 144
v=0 o=GW 2890844527 2890844527 IN IP4 gw1.a.example.com s=- c=IN IP4 gw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 IN IP4 gw1.a.example.com s = -c = In ip4 gw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 100 Trying Proxy 1 -> NGW 1
F4 100プロキシ1-> NGW 1を試します
SIP/2.0 100 Trying
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 600 Busy Everywhere Bob -> Proxy 1
F5 600忙しい場所で忙しいボブ - >プロキシ1
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F6 ACK Proxy 1 -> Bob
F6 ACKプロキシ1->ボブ
ACK bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F7 600 Busy Everywhere Proxy 1 -> NGW 1
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F8 ACK NGW 1 -> Proxy 1
F8 ACK NGW 1->プロキシ1
ACK bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F9 REL NGW 1 -> Alice
f9 rel ngw 1->アリス
REL CauseCode=17 Busy
rel causecode = 17ビジー
F10 RLC Alice -> NGW 1
F10 RLCアリス - > NGW 1
RLC
RLC
Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 600 F5 |
| | |<---------------|
| | | ACK F6 |
| | 600 F7 |--------------->|
| |<---------------| |
| | ACK F8 | |
| ACM F9 |--------------->| |
|<---------------| | |
| One Way Voice | | |
|<===============| | |
| Busy Tone | | |
|<===============| | |
| REL(16) F10 | | |
|--------------->| | |
| RLC F11 | | |
|<---------------| | |
| | | |
In this scenario, Alice calls Bob through Network Gateway NGW 1 and Proxy 1. The call is routed to Bob by Proxy 1. The call is rejected by the Bob client. NGW 1 sets up a two way voice path to Alice and plays busy tone. The caller then disconnects
このシナリオでは、アリスはネットワークゲートウェイNGW 1とプロキシ1を介してボブを呼び出します。コールはプロキシ1によってボブにルーティングされます。コールはボブクライアントによって拒否されます。NGW 1は、アリスへの双方向の音声パスを設定し、忙しいトーンを演奏します。その後、発信者は切断されます
NGW 1 plays the busy tone since the IAM (F1) indicates the interworking is present. In scenario 5.2.2., with no interworking, the busy indication is carried in the REL Cause value and is generated locally instead.
IAM（F1）がインターワーキングが存在することを示しているため、NGW 1は忙しいトーンを再生します。シナリオ5.2.2。では、交流がないため、忙しい兆候はREL原因値で運ばれ、代わりにローカルに生成されます。
Again, note that for ETSI or ITU ISUP, a CONnect message would be sent instead of the Answer Message.
繰り返しますが、ETSIまたはITU ISUPの場合、回答メッセージの代わりに接続メッセージが送信されることに注意してください。
Message Details
メッセージの詳細
F1 IAM Alice -> NGW 1
f1 iam alice-> ngw 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
Interworking=encountered
F2 INVITE NGW1 -> Proxy 1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */
F3 INVITE Proxy 1 -> Bob
F3招待プロキシ1->ボブ
INVITE bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
s=-
c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying Bob -> Proxy 1
F4 100 BOBを試す - >プロキシ1
SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 600 Busy Everywhere Bob -> Proxy 1
F5 600忙しい場所で忙しいボブ - >プロキシ1
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F6 ACK Proxy 1 -> Bob
F6 ACKプロキシ1->ボブ
ACK bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F7 600 Busy Everywhere Proxy 1 -> NGW 1
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F8 ACK NGW 1 -> Proxy 1
F8 ACK NGW 1->プロキシ1
ACK sip:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F9 ACM NGW 1 -> Alice
F9 ACM NGW 1->アリス
ACM
ACM
/* A one way speech path is established between NGW 1 and Alice. */
/* Call Released after Alice hangs up. */
F10 REL Alice -> NGW 1
f10 rel alice-> ngw 1
REL CauseCode=16
rel causecode = 16
F11 RLC NGW 1 -> Alice
F11 RLC NGW 1->アリス
RLC
RLC
Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | INVITE F5 |
| | |--------------->|
| | | INVITE F6 |
| | |--------------->|
| | | INVITE F7 |
| | |--------------->|
| | | INVITE F8 |
| | |--------------->|
| | | INVITE F9 |
| | |--------------->|
| REL F10 | | |
|--------------->| | |
| RLC F11 | | |
|<---------------| | |
| | CANCEL F12 | |
| |--------------->| |
| | 200 F13 | |
| |<---------------| |
Alice calls Bob through NGW 1 and Proxy 1. Proxy 1 re-sends the INVITE after the expiration of SIP timer T1 without receiving any response from Bob. Bob never responds with 180 Ringing or any other response (it is reachable but unresponsive). After the expiration of a timer, Alice's network disconnects the call by sending a Release message REL. The Gateway maps this to a CANCEL.
アリスはNGW 1とプロキシ1を通じてボブに電話をかけます。Proxy1は、BOBから応答を受けずに、SIPタイマーT1の有効期限が切れた後、招待状を再送信します。ボブは180リンギングやその他の応答で応答することはありません（到達可能ですが、反応しません）。タイマーの有効期限が切れた後、AliceのネットワークはリリースメッセージRelを送信することにより、呼び出しを切断します。ゲートウェイはこれをキャンセルにマッピングします。
Message Details
メッセージの詳細
F1 IAM Alice -> NGW 1
f1 iam alice-> ngw 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE Alice -> Proxy 1
F2アリスを招待 - >プロキシ1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */
F3 INVITE Proxy 1 -> Bob
F3招待プロキシ1->ボブ
INVITE sip:bob@b.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0
o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com
c c=IN IP4 ngw1.a.example.com
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F4 100 Trying Proxy 1 -> NGW 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 INVITE Proxy 1 -> Bob
F5招待プロキシ1->ボブ
Same as Message F3
メッセージF3と同じ
F6 INVITE Proxy 1 -> Bob
F6招待プロキシ1->ボブ
Same as Message F3
メッセージF3と同じ
F7 INVITE Proxy 1 -> Bob
F7招待プロキシ1->ボブ
Same as Message F3
メッセージF3と同じ
F8 INVITE Proxy 1 -> Bob
F8招待プロキシ1->ボブ
Same as Message F3
メッセージF3と同じ
F9 INVITE Proxy 1 -> Bob
F9招待プロキシ1->ボブ
Same as Message F3
メッセージF3と同じ
/* Timer expires in Alice's access network. */
F10 REL Alice -> NGW 1
f10 rel alice-> ngw 1
REL CauseCode=16 Normal
rel causecode = 16正常
F11 RLC NGW 1 -> Alice
F11 RLC NGW 1->アリス
RLC F12 CANCEL NGW 1 -> Proxy 1
RLC F12キャンセルNGW 1->プロキシ1
CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 CANCEL
Content-Length: 0
F13 200 OK Proxy 1 -> NGW 1
F13 200 OKプロキシ1-> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 CANCEL
Content-Length: 0
Switch A NGW 1 Stateless Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | INVITE F4 |--------------->|
| |--------------->| INVITE F5 |
| | INVITE F6 |--------------->|
| |--------------->| INVITE F7 |
| | INVITE F8 |--------------->|
| |--------------->| INVITE F9 |
| | INVITE F10 |--------------->|
| |--------------->| INVITE F11 |
| | INVITE F12 |--------------->|
| |--------------->| INVITE F13 |
| | |--------------->|
| REL F14 | | |
|--------------->| | |
| RLC F15 | | |
|<---------------| | |
In this scenario, Alice calls Bob through NGW 1 and Proxy 1. Since Proxy 1 is stateless (it does not send a 100 Trying response), NGW 1 re-sends the INVITE message after the expiration of SIP timer T1. Bob does not respond with 180 Ringing. Alice's network disconnects the call with a release REL (CauseCode=102 Timeout).
このシナリオでは、アリスはNGW 1とプロキシ1を通じてボブを呼び出します。プロキシ1はステートレスであるため（100試行の応答は送信されません）、NGW 1はSIPタイマーT1の有効期限後に招待メッセージを再送信します。ボブは180リンギングで反応しません。Aliceのネットワークは、リリースREL（CauseCode = 102 Timeout）でコールを切断します。
Message Details
メッセージの詳細
F1 IAM Alice -> NGW 1
f1 iam alice-> ngw 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE NGW 1 -> Proxy 1
F2 NGW 1->プロキシ1を招待します
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */
F3 INVITE Proxy 1 -> Bob
F3招待プロキシ1->ボブ
INVITE sip:bob@b.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
Max-Forwards: 69
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 INVITE NGW 1 -> Proxy 1
F4 NGW 1->プロキシ1を招待します
Same as Message F2
メッセージF2と同じ
F5 INVITE Proxy 1 -> Bob
F5招待プロキシ1->ボブ
Same as Message F3 F6 INVITE NGW 1 -> Proxy 1
メッセージF3 f6と同じNGW 1->プロキシ1
Same as Message F2
メッセージF2と同じ
F7 INVITE Proxy 1 -> Bob
F7招待プロキシ1->ボブ
Same as Message F3
メッセージF3と同じ
F8 INVITE NGW 1 -> Proxy 1
F8 NGW 1->プロキシ1を招待します
Same as Message F2
メッセージF2と同じ
F9 INVITE Proxy 1 -> Bob
F9招待プロキシ1->ボブ
Same as Message F3
メッセージF3と同じ
F10 INVITE NGW 1 -> Proxy 1
F10 NGW 1->プロキシ1を招待します
Same as Message F2
メッセージF2と同じ
F11 INVITE Proxy 1 -> Bob
F11招待プロキシ1->ボブ
Same as Message F3
メッセージF3と同じ
F12 INVITE NGW 1 -> Proxy 1
F12 NGW 1->プロキシ1を招待します
Same as Message F2
メッセージF2と同じ
F13 INVITE Proxy 1 -> Bob
f13プロキシ1->ボブを招待します
Same as Message F3
メッセージF3と同じ
/* A timer expires in Alice's access network. */
F14 REL Alice -> NGW 1
f14 rel alice-> ngw 1
REL CauseCode=102 Timeout F15 RLC NGW 1 -> Alice
Rel CauseCode = 102タイムアウトF15 RLC NGW 1-> Alice
RLC
RLC
Switch A NGW 1 Proxy 1 Bob
| | | |
| IAM F1 | | |
|--------------->| INVITE F2 | |
| |--------------->| INVITE F3 |
| | 100 F4 |--------------->|
| |<---------------| |
| | | 180 F5 |
| | 180 F6 |<---------------|
| ACM F7 |<---------------| |
|<---------------| | |
| One Way Voice | | |
|<===============| | |
| Ringing Tone | | |
|<===============| | |
| | | |
| REL F8 | | |
|--------------->| | |
| RLC F9 | | |
|<---------------| CANCEL F10 | |
| |--------------->| |
| | 200 F11 | |
| |<---------------| |
| | | CANCEL F12 |
| | |--------------->|
| | | 200 F13 |
| | |<---------------|
| | | 487 F14 |
| | |<---------------|
| | | ACK F15 |
| | 487 F16 |--------------->|
| |<---------------| |
| | ACK F17 | |
| |--------------->| |
| | | |
In this scenario, Alice calls Bob through NGW 1 and Proxy 1. Bob does not respond with 200 OK. NGW 1 plays ringing tone since the ACM indicates that interworking has been encountered. Alice disconnects the call with a Release message REL which is mapped by NGW 1 to a CANCEL. Note that if Bob had sent a 200 OK response after the REL, NGW 1 would have sent an ACK and then a BYE to properly terminate the call.
このシナリオでは、アリスはNGW 1とプロキシ1を通じてボブを呼び出します。ボブは200 OKで応答しません。NGW 1は、ACMがインターワーキングに遭遇したことを示しているため、リンギングトーンを再生します。Aliceは、NGW 1によってマッピングされたリリースメッセージRelでコールを切断します。BobがRELの後に200 OK応答を送信した場合、NGW 1はACKを送信し、次に通話を適切に終了するためにさようならを送信していたことに注意してください。
Message Details
メッセージの詳細
F1 IAM Alice -> NGW 1
f1 iam alice-> ngw 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=972-555-2222,NPI=E.164,NOA=National
F2 INVITE Alice -> Proxy 1
F2アリスを招待 - >プロキシ1
INVITE sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine a route for
+19725552222. The call is then forwarded to Bob. */
F3 INVITE Proxy 1 -> Bob
F3招待プロキシ1->ボブ
INVITE sip:bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:ngw1@a.example.com;transport=tcp>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844527 2890844527 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844527 2890844527 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 100 Trying Bob -> Proxy 1
F4 100 BOBを試す - >プロキシ1
SIP/2.0 100 Trying
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.201
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 180 Ringing Bob -> Proxy 1
F5 180リンギングボブ - >プロキシ1
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com;transport=tcp>
Content-Length: 0
F6 180 Ringing Proxy 1 -> NGW 1
F6 180リンギングプロキシ1-> NGW 1
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:bob@client.b.example.com>
Content-Length: 0
F7 ACM NGW 1 -> Alice
F7 ACM NGW 1->アリス
ACM
ACM
/* Alice hangs up */
F8 REL Alice -> NGW 1
f8 rel alice-> ngw 1
REL CauseCode=16 Normal
rel causecode = 16正常
F9 RLC NGW 1 -> Alice
F9 RLC NGW 1->アリス
RLC
RLC
F10 CANCEL NGW 1 -> Proxy 1
F10 NGW 1->プロキシ1をキャンセルします
CANCEL sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 CANCEL
Content-Length: 0
F11 200 OK Proxy 1 -> NGW 1
F11 200 OKプロキシ1-> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 CANCEL
Content-Length: 0
F12 CANCEL Proxy 1 -> Bob
f12プロキシ1->ボブをキャンセルします
CANCEL sip:bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 CANCEL
Content-Length: 0
F13 200 OK Bob -> Proxy 1
F13 200 OK BOB->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 CANCEL
Content-Length: 0
F14 487 Request Terminated Bob -> Proxy 1
F14 487リクエスト終了ボブ - >プロキシ1
SIP/2.0 487 Request Terminated
Via: SIP/2.0/TCP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F15 ACK Proxy 1 -> Bob
F15 ACKプロキシ1->ボブ
ACK sip:bob@b.example.com SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F16 487 Request Terminated Proxy 1 -> NGW 1
F16 487リクエスト終了プロキシ1-> NGW 1
SIP/2.0 487 Request Terminated
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
;received=192.0.2.103
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F17 ACK NGW 1 -> Proxy 1
F17 ACK NGW 1->プロキシ1
ACK sip:+19725552222@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TCP ngw1.a.example.com:5060;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sip:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sip:+19725552222@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 4Fde34wkd11wsGFDs3@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
In these scenarios, both the caller and the called party are in the telephone network, either normal PSTN subscribers or PBX extensions. The calls route through two Gateways and at least one SIP Proxy Server. The Proxy Server performs the authentication and location of the Gateways.
これらのシナリオでは、発信者と呼び出された当事者の両方が、通常のPSTNサブスクライバーまたはPBX拡張機能のいずれかで、電話ネットワークにあります。コールルートは、2つのゲートウェイと少なくとも1つのSIPプロキシサーバーを通ります。プロキシサーバーは、ゲートウェイの認証と場所を実行します。
Again it is noted that the intent of this call flows document is not to provide a detailed parameter level mapping of SIP to PSTN protocols. For information on SIP to ISUP mapping, the reader is referred to other references [4].
繰り返しますが、この呼び出しFlowsドキュメントの意図は、SIPのPSTNプロトコルへの詳細なパラメーターレベルマッピングを提供しないことに注意してください。SIPからISUPマッピングの詳細については、読者は他の参照[4]を参照します。
In these scenarios, the call is successfully completed between the two Gateways, allowing the PSTN or PBX users to communicate. The 183 Session Progress response is used to indicate that in-band alerting may flow from the called party telephone switch to the caller.
これらのシナリオでは、2つのゲートウェイ間でコールが正常に完了し、PSTNまたはPBXユーザーが通信できるようになります。183セッションの進行状況応答を使用して、バンド内の警告が発信者に呼び出された電話スイッチから発信者に流れることを示すために使用されます。
Switch A NGW 1 Proxy 1 GW 2 Switch C
| | | | |
| IAM F1 | | | |
|------------->| | | |
| | INVITE F2 | | |
| |------------->| INVITE F3 | |
| | |------------->| IAM F4 |
| | | |------------->|
| | | | ACM F5 |
| | | 183 F6 |<-------------|
| | 183 F7 |<-------------| |
| ACM F8 |<-------------| | |
|<-------------| | | |
| One Way Voice| Two Way RTP Media | One Way Voice|
|<=============|<===========================>|<=============|
| | | | ANM F9 |
| | | 200 F10 |<-------------|
| | 200 F11 |<-------------| |
| ANM F12 |<-------------| | |
|<-------------| | | |
| | ACK F13 | | |
| |------------->| ACK F14 | |
| | |------------->| |
|Both Way Voice| Both Way RTP Media |Both Way Voice|
|<=============|<===========================>|<=============|
| | | | REL F15 |
| | | |<-------------|
| | | BYE F16 | |
| | BYE F18 |<-------------| RLC F17 |
| |<-------------| |------------->|
| | | | |
| | 200 F19 | | |
| |------------->| 200 F20 | |
| | |------------->| |
| REL F21 | | | |
|<-------------| | | |
| RLC F22 | | | |
|------------->| | | |
| | | | |
In this scenario, Alice in the PSTN calls Carol who is an extension on a PBX. Alice's telephone switch signals via SS7 to the Network Gateway NGW 1, while Carol's PBX signals via SS7 with the Gateway GW 2. The CdPN and CgPN are mapped by GW 1 into SIP URIs and placed in the To and From headers. Proxy 1 looks up the dialed digits in the Request-URI and maps the digits to the PBX extension of Carol, which is served by GW 2. The Proxy in F3 uses the host portion of the Request-URI to identify what private dialing plan is being referenced. The INVITE is then forwarded to GW 2 for call completion. An early media path is established end-to-end so that Alice can hear the ringing tone generated by PBX C.
このシナリオでは、PSTNのアリスは、PBXの拡張機能であるキャロルを呼び出します。Aliceの電話スイッチはSS7を介してネットワークゲートウェイNGW 1に、CarolのPBXはGateway GW 2を介してSS7を介してシグナルを与えます。CDPNとCGPNは、GW 1によってSIP URIにマッピングされ、Headersに配置されます。Proxy 1は、リクエスト-URIのダイヤル桁を調べ、GW 2が提供するキャロルのPBX拡張機能に数字をマップします。F3のプロキシは、リクエスト-URIのホスト部分を使用して、プライベートダイヤルプランと識別します。参照される。招待状は、コール完了のためにGW 2に転送されます。AliceがPBX Cによって生成されたリンギングトーンを聞くことができるように、初期のメディアパスがエンドツーエンドで確立されます。
Carol answers the call and the media path is cut through in both directions. Bob hangs up terminating the call.
キャロルは電話に応答し、メディアパスは両方向に切り抜かれます。ボブは電話を切って電話をかけます。
Message Details
メッセージの詳細
F1 IAM Switch Alice -> NGW 1
F1 IAMスイッチアリス - > NGW 1
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=918-555-3333,NPI=E.164,NOA=National
F2 INVITE NGW 1 -> Proxy 1
F2 NGW 1->プロキシ1を招待します
INVITE sips:+19185553333@ss1.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
Max-Forwards: 70
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sips:ngw1@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844526 2890844526 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844526 2890844526 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 consults Location Service and translates the dialed number
to a private number in the Request-URI*/
F3 INVITE Proxy 1 -> GW 2
F3招待プロキシ1-> GW 2
INVITE sips:4443333@gw2.a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKwqwee65
;received=192.0.2.103
Max-Forwards: 69
Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sips:+19185553333@ss1.a.example.com;user=phone>
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sips:ngw1@a.example.com>
Content-Type: application/sdp
Content-Length: 146
v=0 o=GW 2890844526 2890844526 IN IP4 ngw1.a.example.com s=- c=IN IP4 ngw1.a.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844526 2890844526 in ip4 ngw1.a.example.com s = - c = in ip4 ngw1.a.example.com t = 0 0 m = audio 3456 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 IAM GW 2 -> Switch C
F4 IAM GW 2->スイッチc
IAM
CgPN=314-555-1111,NPI=E.164,NOA=National
CdPN=444-3333,NPI=Private,NOA=Subscriber
F5 ACM Switch C -> GW 2
F5 ACMスイッチC-> GW 2
ACM
ACM
/* Based on the ACM message, GW 2 returns a 183 response. In-band
call progress indications are sent to Alice through NGW 1. */
F6 183 Session Progress GW 2 -> Proxy 1
F6 183セッションの進捗GW 2->プロキシ1
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sips:4443333@gw2.a.example.com>
Content-Type: application/sdp
Content-Length: 143
v=0 o=GW 987654321 987654321 IN IP4 gw2.a.example.com s=- c=IN IP4 gw2.a.example.com t=0 0 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 987654321 987654321 IN IP4 gw2.a.example.com s = -c = In ip4 gw2.a.example.com t = 0 0 m = audio 14918 rtp/avp 0 a = rtpmap：0 pcmu/8000
F7 183 Session Progress Proxy 1 -> GW 1
F7 183セッション進行プロキシ1-> GW 1
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sips:4443333@gw2.a.example.com>
Content-Type: application/sdp
Content-Length: 143
v=0 o=GW 987654321 987654321 IN IP4 gw2.a.example.com s=- c=IN IP4 gw2.a.example.com t=0 0 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 987654321 987654321 IN IP4 gw2.a.example.com s = -c = In ip4 gw2.a.example.com t = 0 0 m = audio 14918 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* NGW 1 receives packets from GW 2 with encoded ringback, tones or
other audio. NGW 1 decodes this and places it on the originating
trunk. */
F8 ACM NGW 1 -> Switch A
F8 ACM NGW 1->スイッチa
ACM
ACM
/* Bob answers */
F9 ANM Switch C -> GW 2
ANM
ANM
F10 200 OK GW 2 -> Proxy 1
F10 200 OK GW 2->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sips:4443333@gw2.a.example.com>
Content-Type: application/sdp
Content-Length: 143
v=0 o=GW 987654321 987654321 IN IP4 gw2.a.example.com s=- c=IN IP4 gw2.a.example.com t=0 0 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 987654321 987654321 IN IP4 gw2.a.example.com s = -c = In ip4 gw2.a.example.com t = 0 0 m = audio 14918 rtp/avp 0 a = rtpmap：0 pcmu/8000
F11 200 OK Proxy 1 -> NGW 1
F11 200 OKプロキシ1-> NGW 1
SIP/2.0 200 OK
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103
Record-Route: <sips:ss1.a.example.com;lr>
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 1 INVITE
Contact: <sips:4443333@gw2.a.example.com>
Content-Type: application/sdp
Content-Length: 143
v=0
o=GW 987654321 987654321 IN IP4 gw2.a.example.com
s=-
c=IN IP4 gw2.a.example.com
t=0 0
m=audio 14918 RTP/AVP 0
a=rtpmap:0 PCMU/8000
F12 ANM NGW 1 -> Switch A
F12 ANM NGW 1->スイッチa
ANM
ANM
F13 ACK NGW 1 -> Proxy 1
F13 ACK NGW 1->プロキシ1
ACK sips:4443333@gw2.a.example.com SIP/2.0
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F14 ACK Proxy 1 -> GW 2
F14 ACKプロキシ1-> GW 2
ACK sips:4443333@gw2.a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS ngw1.a.example.com:5061;branch=z9hG4bKlueha2
;received=192.0.2.103
Max-Forwards: 69
From: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
To: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
/* RTP streams are established between NGW 1 and GW 2. */
/* Bob Hangs Up with Alice. */
F15 REL Switch C -> GW 2
F15 RELスイッチC-> GW 2
REL CauseCode=16 Normal F16 BYE GW 2 -> Proxy 1
rel causecode = 16通常のf16バイGW 2->プロキシ1
BYE sips:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
Max-Forwards: 70
Route: <sips:ss1.a.example.com;lr>
From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 4 BYE
Content-Length: 0
F17 RLC GW 2 -> Switch C
F17 RLC GW 2->スイッチc
RLC
RLC
F18 BYE Proxy 1 -> NGW 1
F18 Bye Proxy 1-> ngw 1
BYE sips:ngw1@a.example.com SIP/2.0
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
;received=192.0.2.202
Max-Forwards: 69
From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 4 BYE
Content-Length: 0
F19 200 OK NGW 1 -> Proxy 1
F19 200 OK NGW 1->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/TLS ss1.a.example.com:5061;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
;received=192.0.2.202
From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 4 BYE
Content-Length: 0
F20 200 OK Proxy 1 -> GW 2
SIP/2.0 200 OK
Via: SIP/2.0/TLS gw2.a.example.com:5061;branch=z9hG4bKtexx6
;received=192.0.2.202
From: <sips:+19185553333@ss1.a.example.com;user=phone>;tag=314159
To: <sips:+13145551111@ngw1.a.example.com;user=phone>;tag=7643kals
Call-ID: 2xTb9vxSit55XU7p8@ngw1.a.example.com
CSeq: 4 BYE
Content-Length: 0
F21 REL Switch C -> GW 2
F21 RELスイッチC-> GW 2
REL CauseCode=16 Normal
rel causecode = 16正常
F22 RLC GW 2 -> Switch C
F22 RLC GW 2->スイッチc
RLC
RLC
PBX A GW 1 Proxy 1 GW 2 GW 3 PBX C
| | | | | |
| Seizure | | | | |
|----------->| | | | |
| Wink | | | | |
|<-----------| | | | |
|MF Digits F1| | | | |
|----------->| | | | |
| | INVITE F2 | | | |
| |----------->| INVITE F3 | | |
| | |----------->| | |
| | | 503 F4 | | |
| | |<-----------| | |
| | | ACK F5 | | |
| | |----------->| | |
| | | INVITE F6 | |
| | |------------------------>| SETUP F7 |
| | | 100 F8 |----------->|
| | |<------------------------|CALL PROC F9|
| | | |<-----------|
| | | | ALERT F10 |
| | | 180 F11 |<-----------|
| | 180 F12 |<------------------------| |
| |<-----------| | |
| Ringtone | | |OneWay Voice|
|<===========| | |<===========|
| | | | CONNect F13|
| | | 200 F14 |<-----------|
| | 200 F15 |<------------------------| |
| Seizure |<-----------| | |
|<-----------| ACK F16 | | |
| |----------->| ACK F17 | |
| | |------------------------>|CONN ACK F18|
| | | |----------->|
|BothWayVoice| Both Way RTP Media |BothWayVoice|
|<==========>|<====================================>|<==========>|
| | | | DISC F19 |
| | | |<-----------|
| | | BYE F20 | |
| | BYE F21 |<------------------------| REL F22 |
|Seiz Removal|<-----------| |----------->|
|<-----------| 200 F23 | | |
|Seiz Removal|----------->| 200 F24 | |
|----------->| |------------------------>| REL COM F25|
| | | |<-----------|
| | | | |
PBX Alice calls PBX Carol via Gateway GW 1 and Proxy 1. During the attempt to reach Carol via GW 2, an error is encountered - Proxy 1 receives a 503 Service Unavailable (F4) response to the forwarded INVITE. This could be due to all circuits being busy, or some other outage at GW 2. Proxy 1 recognizes the error and uses an alternative route via GW 3 to terminate the call. From there, the call proceeds normally with Carol answering the call. The call is terminated when Carol hangs up.
PBX Aliceは、Gateway GW 1およびProxy 1を介してPBX Carolを呼び出します。GW2を介してキャロルに到達しようとする試みで、エラーが発生します。これは、すべての回路が忙しいこと、またはGW 2での他の停止によるものである可能性があります。Proxy1はエラーを認識し、GW 3を介して代替ルートを使用してコールを終了します。そこから、コールは通常、Carolが電話に応答すると進行します。キャロルが電話を切るときに呼び出しが終了します。
Message Details
メッセージの詳細
PBX Alice -> GW 1
PBXアリス - > GW 1
Seizure
発作
GW 1 -> PBX A
GW 1-> PBX a
Wink
ウィンク
F1 MF Digits PBX Alice -> GW 1
f1 mf digits pbx alice-> gw 1
KP 444 3333 ST
KP 444 3333 ST
F2 INVITE GW 1 -> Proxy 1
F2 GW 1->プロキシ1を招待します
INVITE sip:4443333@ss1.a.example.com SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:551313@gw1.a.example.com>
Content-Type: application/sdp
Content-Length: 155
v=0 o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com s=- c=IN IP4 gw1.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844526 2890844526 in ip4 gw1.a.example.com s = -c = in ip4 gw1.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
/* Proxy 1 uses a Location Service function to determine where B is
located. Response is returned listing alternative routes, GW2 and
GW3, which are then tried sequentially. */
F3 INVITE Proxy 1 -> GW 2
F3招待プロキシ1-> GW 2
INVITE sip:4443333@gw2.a.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:551313@gw1.a.example.com>
Content-Type: application/sdp
Content-Length: 155
v=0 o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com s=- c=IN IP4 gw1.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844526 2890844526 in ip4 gw1.a.example.com s = -c = in ip4 gw1.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F4 503 Service Unavailable GW 2 -> Proxy 1
F4 503サービス利用不能GW 2->プロキシ1
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F5 ACK Proxy 1 -> GW 2
F5 ACKプロキシ1-> GW 2
ACK sip:4443333@ss1.a.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.1
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forward: 70
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=314159
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F6 INVITE Proxy 1 -> GW 3
F6招待プロキシ1-> GW 3
INVITE sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:551313@gw1.a.example.com>
Content-Type: application/sdp
Content-Length: 155
v=0 o=GW 2890844526 2890844526 IN IP4 gw1.a.example.com s=- c=IN IP4 gw1.a.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 2890844526 2890844526 in ip4 gw1.a.example.com s = -c = in ip4 gw1.a.example.com t = 0 0 m = audio 49172 rtp/avp 0 a = rtpmap：0 pcmu/8000
F7 SETUP GW 3 -> PBX C
F7セットアップGW 3-> PBX c
Protocol discriminator=Q.931
Message type=SETUP
Bearer capability: Information transfer capability=0 (Speech) or 16
(3.1 kHz audio)
Channel identification=Preferred or exclusive B-channel
Progress indicator=1 (Call is not end-to-end ISDN; further call
progress information may be available inband)
Called party number:
Type of number and numbering plan ID=33 (National number in ISDN
numbering plan)
Digits=918-555-3333
F8 100 Trying GW 3 -> Proxy 1
SIP/2.0 100 Trying
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE
Content-Length: 0
F9 CALL PROCeeding PBX C -> GW 3
f9コールPBX C-> GW 3
Protocol discriminator=Q.931 Message type=CALL PROC
プロトコル識別子= Q.931メッセージタイプ= call proc
F10 ALERT PBX C -> GW 3
F10アラートPBX C-> GW 3
Protocol discriminator=Q.931 Message type=PROG
プロトコル識別子= Q.931メッセージタイプ= prog
/* Based on ALERT message, GW 3 returns a 180 response. */
F11 180 Ringing GW 3 -> Proxy 1
F11 180リンギングGW 3->プロキシ1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
Content-Length: 0
F12 180 Ringing Proxy 1 -> GW 1
F12 180リンギングプロキシ1-> GW 1
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
Content-Length: 0
F13 CONNect PBX C -> GW 3
F13接続PBX C-> GW 3
Protocol discriminator=Q.931 Message type=CONN
プロトコル識別子= Q.931メッセージタイプ= conn
F14 200 OK GW 3 -> Proxy 1
F14 200 OK GW 3->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
Content-Type: application/sdp
Content-Length: 143
v=0 o=GW 987654321 987654321 IN IP4 gw3.a.example.com s=- c=IN IP4 gw3.a.example.com t=0 0 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 987654321 987654321 IN IP4 GW3.A.EXAMPLE.COM S = -C = IN IP4 GW3.A.EXAMPLE.COM T = 0 0 M = Audio 14918 RTP/AVP 0 A = RTPMAP：0 PCMU/8000
F15 200 OK Proxy 1 -> GW 1
F15 200 OKプロキシ1-> GW 1
SIP/2.0 200 OK
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Record-Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 INVITE
Contact: <sip:+19185553333@gw3.a.example.com;user=phone>
Content-Type: application/sdp
Content-Length: 143
v=0 o=GW 987654321 987654321 IN IP4 gw3.a.example.com s=- c=IN IP4 gw3.a.example.com t=0 0 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000
V = 0 O = GW 987654321 987654321 IN IP4 GW3.A.EXAMPLE.COM S = -C = IN IP4 GW3.A.EXAMPLE.COM T = 0 0 M = Audio 14918 RTP/AVP 0 A = RTPMAP：0 PCMU/8000
GW 1 -> PBX A
GW 1-> PBX a
Seizure
発作
F16 ACK GW 1 -> Proxy 1
F16 ACK GW 1->プロキシ1
ACK sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F17 ACK Proxy 1 -> GW 3
F17 ACKプロキシ1-> GW 3
ACK sip:+19185553333@gw3.a.example.com;user=phone SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP gw1.a.example.com:5060;branch=z9hG4bKwqwee65
;received=192.0.2.201
Max-Forwards: 69
From: <sip:551313@gw1.a.example.com>;tag=63412s
To: <sip:4443333@ss1.a.example.com>;tag=123456789
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 ACK
Content-Length: 0
F18 CONNect ACK GW 3 -> PBX C
Protocol discriminator=Q.931 Message type=CONN ACK
プロトコル差別= Q.931メッセージタイプ= conn ack
/* RTP streams are established between GW 1 and GW 3. */
/* Bob Hangs Up with Alice. */
F19 DISConnect PBX C -> GW 3
F19 PBX C-> GW 3を切断します
Protocol discriminator=Q.931 Message type=DISC Cause=16 (Normal clearing)
プロトコル識別子= Q.931メッセージタイプ=ディスク原因= 16（通常のクリアリング）
F20 BYE GW 3 -> Proxy 1
F20バイGW 3->プロキシ1
BYE sip:551313@gw1.a.example.com SIP/2.0
Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
Max-Forwards: 70
Route: <sip:ss1.a.example.com;lr>
From: <sip:4443333@ss1.a.example.com>;tag=123456789
To: <sip:551313@gw1.a.example.com>;tag=63412s
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 BYE
Content-Length: 0
F21 BYE Proxy 1 -> GW 1
F21バイプロキシ1-> GW 1
BYE sip:551313@gw1.a.example.com SIP/2.0
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
;received=192.0.2.203
Max-Forwards: 69
From: <sip:4443333@ss1.a.example.com>;tag=123456789
To: <sip:551313@gw1.a.example.com>;tag=63412s
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 BYE
Content-Length: 0
GW 1 -> PBX A
GW 1-> PBX a
Seizure removal F22 RELease GW 3 -> PBX C
発作除去F22リリースGW 3-> PBX C
Protocol discriminator=Q.931 Message type=REL
プロトコル識別子= Q.931メッセージタイプ= rel
F23 200 OK GW 1 -> Proxy 1
F23 200 OK GW 1->プロキシ1
SIP/2.0 200 OK
Via: SIP/2.0/UDP ss1.a.example.com:5060;branch=z9hG4bK2d4790.2
;received=192.0.2.111
Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
;received=192.0.2.203
From: <sip:4443333@ss1.a.example.com>;tag=123456789
To: <sip:551313@gw1.a.example.com>;tag=63412s
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 BYE
Content-Length: 0
F24 200 OK Proxy 1 -> GW 3
F24 200 OKプロキシ1-> GW 3
SIP/2.0 200 OK
Via: SIP/2.0/UDP gw3.a.example.com:5060;branch=z9hG4bKkdjuwq
;received=192.0.2.203
From: <sip:4443333@ss1.a.example.com>;tag=123456789
To: <sip:551313@gw1.a.example.com>;tag=63412s
Call-ID: 2xTb9vxSit55XU7p8@gw1.a.example.com
CSeq: 1 BYE
Content-Length: 0
F25 RELease COMplete PBX C -> GW 3
F25リリース完全なPBX C-> GW 3
Protocol discriminator=Q.931 Message type=REL COM
プロトコル識別子= Q.931メッセージタイプ= rel com
PBX Alice -> GW 1
PBXアリス - > GW 1
Seizure removal
発作除去
This document provides examples of mapping from SIP to ISUP and ISUP to SIP. The gateways in these examples are compliant with the Security Considerations Section of RFC 3398 [4] which is summarized here.
このドキュメントには、SIPからISUPへのマッピングの例とSIPへのISUPの例を提供します。これらの例のゲートウェイは、RFC 3398 [4]のセキュリティに関する考慮事項セクションに準拠しています。これはここにまとめられています。
There are few security concerns relating to the mapping of ISUP to SIP besides privacy considerations in the calling party number passing. Some concerns relating to the mapping from tel URI parameters to ISUP include the user creation of parameters and codes relating to called number and local number portability (LNP). An operator of a gateway should use policies similar to those present in PSTN switches to avoid security problems.
呼び出されたパーティー番号の合格におけるプライバシーに関する考慮事項に加えて、ISUPのマッピングに関連するセキュリティ上の懸念はほとんどありません。Tel URIパラメーターからISUPへのマッピングに関連するいくつかの懸念には、呼び出された番号とローカル番号ポータビリティ（LNP）に関連するパラメーターとコードのユーザー作成が含まれます。ゲートウェイのオペレーターは、セキュリティの問題を回避するために、PSTNスイッチに存在するものと同様のポリシーを使用する必要があります。
The mapping from a SIP response code to an ISUP Cause Code presents a theoretical risk, so a gateway operator may implement policies controlling this mapping. Gateways should also not rely on the contents of the From header field for identity information, as it may be arbitrarily populated by a user. Instead, some sort of cryptographic authentication and authorization should be used for identity determination. These flows show both HTTP Digest for authentication of users, although for brevity, the challenge is not always shown.
SIP応答コードからISUP原因コードへのマッピングは、理論的リスクを提示するため、ゲートウェイオペレーターはこのマッピングを制御するポリシーを実装できます。また、ゲートウェイは、ユーザーがarbitrarily意的に入力される可能性があるため、ID情報のHeaderフィールドの内容に依存してはなりません。代わりに、ある種の暗号化認証と承認をアイデンティティの決定に使用する必要があります。これらのフローは、ユーザーの認証のためのHTTPダイジェストの両方を示していますが、簡潔にするために、課題は常に示されているわけではありません。
The early media cut-through shown in some flows is another potential security risk, but it is also required for proper interaction with the PSTN. Again, a gateway operator should use proper policies relating to early media to prevent fraud and misuse. Finally, a user agent (even a properly authenticated one) can launch multiple simultaneous requests through a gateway, constituting a denial of service attack. The adoption of policies to limit the number of simultaneous requests from a single entity may be used to prevent this attack.
いくつかのフローに示されている初期のメディアカットスルーは、別の潜在的なセキュリティリスクですが、PSTNとの適切な相互作用にも必要です。繰り返しますが、ゲートウェイオペレーターは、詐欺や誤用を防ぐために、初期のメディアに関連する適切なポリシーを使用する必要があります。最後に、ユーザーエージェント（適切に認証されたエージェントでも）は、ゲートウェイを介して複数の同時リクエストを起動し、サービス拒否攻撃を構成することができます。単一のエンティティからの同時リクエストの数を制限するためのポリシーの採用は、この攻撃を防ぐために使用できます。
As discussed in the SIP-T framework [7], SIP/ISUP interworking can be employed as an interdomain signaling mechanism that may be subject to pre-existing trust relationships between administrative domains. Any administrative domain implementing SIP-T or SIP/ISUP interworking should have an adequate security apparatus (including elements that manage any appropriate policies to manage fraud and billing in an interdomain environment) in place to ensure that the translation of ISUP information does not result in any security violations.
SIP-Tフレームワーク[7]で説明したように、SIP/ISUPインターワーキングは、管理ドメイン間の既存の信頼関係の影響を受ける可能性のあるドメイン間シグナル伝達メカニズムとして採用できます。SIP-TまたはSIP/ISUPインターワーキングを実装する管理ドメインには、ISUP情報の翻訳が結果として生じることを保証するために、適切な詐欺と請求を管理するための適切なポリシーを管理する要素を含む）を備えている必要があります。セキュリティ違反。
Although no examples of this are shown in this document, transporting ISUP in SIP bodies may provide opportunities for abuse, fraud, and privacy concerns, especially when SIP-T requests can be generated, inspected or modified by arbitrary SIP endpoints. ISUP MIME bodies should be secured (preferably with S/MIME as detailed in RFC 3261 [2]) to alleviate this concern. Authentication properties provided by S/MIME would allow the recipient of a SIP-T message to ensure that the ISUP MIME body was generated by an authorized entity. Encryption would ensure that only carriers possessing a particular decryption key are capable of inspecting encapsulated ISUP MIME bodies in a SIP request.
このドキュメントにはこれの例は示されていませんが、SIP団体でのISUPの輸送は、特にSIP-Tのリクエストを任意のSIPエンドポイントによって生成、検査、または変更できる場合、虐待、詐欺、プライバシーの懸念の機会を提供する可能性があります。この懸念を軽減するには、isup mime bodyを保護する必要があります（できれば、RFC 3261 [2]で詳述されているS/MIMEで）。S/MIMEが提供する認証プロパティは、SIP-Tメッセージの受信者に、ISUP MIMEボディが認定エンティティによって生成されるようにすることができます。暗号化は、特定の復号化キーを所有するキャリアのみが、SIPリクエストでカプセル化されたISUP MIME体を検査できるようにすることを保証します。
[1] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997.
[1] Bradner、S。、「要件レベルを示すためにRFCで使用するためのキーワード」、BCP 14、RFC 2119、1997年3月。
[2] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M. E. and Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002.
[2] Rosenberg、J.、Schulzrinne、H.、Camarillo、G.、Johnston、A.、Peterson、J.、Sparks、R.、Handley、M。E. and Schooler、「SIP：SESSION INIATIANG Protocol "、RFC 3261、2002年6月。
[3] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with the Session Description Protocol (SDP)", RFC 3264, June 2002.
[3] Rosenberg、J。およびH. Schulzrinne、「セッション説明プロトコル（SDP）のオファー/回答モデル」、RFC 3264、2002年6月。
[4] Camarillo, G., Roach, A. B., Peterson, J. and L. Ong, "Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping", RFC 3398, December 2002.
[4] Camarillo、G.、Roach、A。B.、Peterson、J。and L. Ong、「Integrated Services Digital Network（ISDN）ユーザーパーツ（ISUP）セッション開始プロトコル（SIP）マッピング」、RFC 3398、2002年12月。
[5] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S., Leach, P., Luotonen, A. and L. Stewart, "HTTP Authentication: Basic and Digest Access Authentication", RFC 2617, June 1999.
[5] Franks、J.、Hallam-Baker、P.、Hostetler、J.、Lawrence、S.、Leach、P.、Luotonen、A。and L. Stewart、「HTTP認証：基本および消化アクセス認証」、RFC 2617、1999年6月。
[6] Vaha-Sipila, A., "URLs for Telephone Calls", RFC 2806, April 2000.
[6] Vaha-Sipila、A。、「電話のためのURL」、RFC 2806、2000年4月。
[7] Vemuri, A. and J. Peterson, "Session Initiation Protocol for Telephones (SIP-T): Context and Architectures", BCP 63, RFC 3372, September 2002.
[7] Vemuri、A。およびJ. Peterson、「電話のセッション開始プロトコル（SIP-T）：コンテキストとアーキテクチャ」、BCP 63、RFC 3372、2002年9月。
[8] Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet, F., Watson, M. and M. Zonoun, "MIME media types for ISUP and QSIG Objects", RFC 3204, December 2001.
[8] Zimmerer、E.、Peterson、J.、Vemuri、A.、Ong、L.、Audet、F.、Watson、M.、M。Zonoun、「ISUPおよびQSIGオブジェクトのMIMEメディアタイプ」、RFC 3204、2001年12月。
[9] Faltstrom, P., "E.164 number and DNS", RFC 2916, September 2000.
[9] Faltstrom、P。、「E.164番号とDNS」、RFC 2916、2000年9月。
[10] Johnston, A., Donovan, S., Sparks, R., Cunningham, C. and K. Summers, "Session Initiation Protocol (SIP) Basic Call Flow Examples", RFC 3665, December 2003.
[10] Johnston、A.、Donovan、S.、Sparks、R.、Cunningham、C。and K. Summers、「セッション開始プロトコル（SIP）基本的なコールフローの例」、RFC 3665、2003年12月。
Thanks to Rohan Mahy, Adam Roach, Gonzalo Camarillo, Cullen Jennings, and Tom Taylor for their detailed comments during the final review. Thanks to Dean Willis for his early contributions to the development of this document. Thanks to Jon Peterson for his help on the security section.
Rohan Mahy、Adam Roach、Gonzalo Camarillo、Cullen Jennings、およびTom Taylorの最終レビュー中の詳細なコメントに感謝します。この文書の開発への初期の貢献について、ディーン・ウィリスに感謝します。セキュリティセクションで彼の助けをしてくれたJon Petersonに感謝します。
The authors wish to thank Kundan Singh for performing parser validation of messages.
著者は、メッセージのパーサー検証を実行してくれたKundan Singhに感謝したいと考えています。
The authors wish to thank the following individuals for their participation in a detailed review of this call flows document: Aseem Agarwal, Rafi Assadi, Ben Campbell, Sunitha Kumar, Jon Peterson, Marc Petit-Huguenin, Vidhi Rastogi, and Bodgey Yin Shaohua.
著者は、Aseem Agarwal、Rafi Assadi、Ben Campbell、Sunitha Kumar、Jon Peterson、Marc Petit-Huguenin、Vidhi rastogi、およびBodgey Yin Shaohuaの詳細なレビューに参加してくれたことに感謝します。
The authors also wish to thank the following individuals for their assistance: Jean-Francois Mule, Hemant Agrawal, Henry Sinnreich, David Devanatham, Joe Pizzimenti, Matt Cannon, John Hearty, the whole MCI WorldCom IPOP Design team, Scott Orton, Greg Osterhout, Pat Sollee, Doug Weisenberg, Danny Mistry, Steve McKinnon, and Denise Ingram, Denise Caballero, Tom Redman, Ilya Slain, Pat Sollee, John Truetken, and others from MCI WorldCom, 3Com, Cisco, Lucent and Nortel.
著者はまた、ジャン・フランソワ・ラバ、ヘマント・アグラワル、ヘンリー・シン・デヴァナタム、ジョー・ピジメンティ、マット・キャノン、ジョン・ハーティ、マック・ワールドコム・イポップ・デザインチーム、スコット・オートン、グレッグ・オステルハウト、パット・ソルリー、ダグ・ワイゼンバーグ、ダニー・ミストリー、スティーブ・マッキノン、デニス・イングラム、デニス・カバレロ、トム・レッドマン、イリヤ・スレイン、パット・ソルリー、ジョン・トルエットケン、その他MCI Worldcom、3com、Cisco、Lucent、Nortelのその他。
The IETF takes no position regarding the validity or scope of any intellectual property or other rights that might be claimed to pertain to the implementation or use of the technology described in this document or the extent to which any license under such rights might or might not be available; neither does it represent that it has made any effort to identify any such rights. Information on the IETF's procedures with respect to rights in standards-track and standards-related documentation can be found in BCP-11. Copies of claims of rights made available for publication and any assurances of licenses to be made available, or the result of an attempt made to obtain a general license or permission for the use of such proprietary rights by implementors or users of this specification can be obtained from the IETF Secretariat.
IETFは、知的財産またはその他の権利の有効性または範囲に関して、この文書に記載されているテクノロジーの実装または使用に関連すると主張される可能性のある他の権利、またはそのような権利に基づくライセンスがどの程度であるかについての程度に関連する可能性があるという立場はありません。利用可能;また、そのような権利を特定するために努力したことも表明していません。標準トラックおよび標準関連のドキュメントの権利に関するIETFの手順に関する情報は、BCP-11に記載されています。出版のために利用可能にされた権利の請求のコピーと、利用可能になるライセンスの保証、またはこの仕様の実装者またはユーザーによるそのような独自の権利の使用のための一般的なライセンスまたは許可を取得しようとする試みの結果を得ることができますIETF事務局から。
The IETF invites any interested party to bring to its attention any copyrights, patents or patent applications, or other proprietary rights which may cover technology that may be required to practice this standard. Please address the information to the IETF Executive Director.
IETFは、関心のある当事者に、この基準を実践するために必要な技術をカバーする可能性のある著作権、特許、または特許出願、またはその他の独自の権利を注意深く招待するよう招待しています。情報をIETFエグゼクティブディレクターに宛ててください。
All listed authors actively contributed large amounts of text to this document.
リストされているすべての著者は、このドキュメントに大量のテキストを積極的に提供しました。
Alan Johnston MCI 100 South 4th Street St. Louis, MO 63102 USA
アラン・ジョンストン・マクイ100サウス4番街セントルイス、ミズーリ州63102 USA
EMail: alan.johnston@mci.com
Steve Donovan dynamicsoft, Inc. 5100 Tennyson Parkway Suite 1200 Plano, Texas 75024 USA
Steve Donovan Dynamicsoft、Inc。5100 Tennyson Parkway Suite 1200 Plano、Texas 75024 USA
EMail: sdonovan@dynamicsoft.com
Robert Sparks dynamicsoft, Inc. 5100 Tennyson Parkway Suite 1200 Plano, Texas 75024 USA
Robert Sparks Dynamicsoft、Inc。5100 Tennyson Parkway Suite 1200 Plano、Texas 75024 USA
EMail: rsparks@dynamicsoft.com
Chris Cunningham dynamicsoft, Inc. 5100 Tennyson Parkway Suite 1200 Plano, Texas 75024 USA
Chris Cunningham Dynamicsoft、Inc。5100 Tennyson Parkway Suite 1200 Plano、Texas 75024 USA
EMail: ccunningham@dynamicsoft.com
Kevin Summers Sonus 1701 North Collins Blvd, Suite 3000 Richardson, TX 75080 USA
ケビン・サマーズソヌス1701ノースコリンズブルバード、スイート3000リチャードソン、テキサス75080 USA
EMail: kevin.summers@sonusnet.com
Copyright (C) The Internet Society (2003). All Rights Reserved.
Copyright（c）The Internet Society（2003）。無断転載を禁じます。
This document and translations of it may be copied and furnished to others, and derivative works that comment on or otherwise explain it or assist in its implementation may be prepared, copied, published and distributed, in whole or in part, without restriction of any kind, provided that the above copyright notice and this paragraph are included on all such copies and derivative works. However, this document itself may not be modified in any way, such as by removing the copyright notice or references to the Internet Society or other Internet organizations, except as needed for the purpose of developing Internet standards in which case the procedures for copyrights defined in the Internet Standards process must be followed, or as required to translate it into languages other than English.
このドキュメントと翻訳は他の人にコピーされて提供される場合があります。また、それについてコメントまたは説明する派生作品、またはその実装を支援することは、いかなる種類の制限なしに、準備、コピー、公開、および部分的に配布される場合があります。、上記の著作権通知とこの段落がそのようなすべてのコピーとデリバティブ作品に含まれている場合。ただし、このドキュメント自体は、インターネット協会や他のインターネット組織への著作権通知や参照を削除するなど、いかなる方法でも変更できない場合があります。インターネット標準プロセスに従うか、英語以外の言語に翻訳するために必要な場合に従う必要があります。
The limited permissions granted above are perpetual and will not be revoked by the Internet Society or its successors or assignees.
上記の限られた許可は永続的であり、インターネット社会やその後継者または譲受人によって取り消されることはありません。
This document and the information contained herein is provided on an "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
この文書と本書に含まれる情報は、「現状」に基づいて提供されており、インターネット社会とインターネットエンジニアリングタスクフォースは、ここにある情報の使用が行われないという保証を含むがこれらに限定されないすべての保証を否認します。特定の目的に対する商品性または適合性の権利または黙示的な保証を侵害します。
Acknowledgement
謝辞
Funding for the RFC Editor function is currently provided by the Internet Society.
RFCエディター機能の資金は現在、インターネット協会によって提供されています。