SIP プロバイダのトランク設定情報と質問表

When working with your SIP provider to set up and configure your external trunk in PureCloud, you need to provide them with information about PureCloud. You also need to ask specific questions about configuration options. Once you have your initial trunk configuration in place, youwant to run some tests to make sure that everything is set up properly.

In this article, we provide you with  a list of PureCloud requirements. We also provide you  a list of the RFCs currently supported in PureCloud. You can use this information to assist you with a discussion with your SIP provider.

Also in this article, we provide you with a questionnaire designed to help you discuss your needs and requirements with your SIP provider. We also provide you with a test plan designed to help you make sure that your SIP trunk is configured properly.


Discuss these requirements with your SIP provider.

  • 複数の Edge を展開する場合は、順次トランク フェイルオーバーが推奨されます。
  • キャリアは、Edge サーバーの IP アドレスからシグナリングと音声トラフィック(SIP/RTP)を許可する必要があります。
  • 推奨された SIP オプションは、トランクのフェールオーバーがないか非対応の Edges を検出するために ping を打ちます。

  • すべてのインバウンド/アウトバウンド トラフィックの SIP のポート 5060
  • 帯域外の DTMF (RFC 2833)
  • g.711 パススルーでファックス
  • ANI あり/なしのインバウンド コール
  • ANI ありのアウトバウンド コール
  • 初期メディア
  • 着信コール転送
  • 応答不可 (キャリアから 503 の応答)
  • List IP addresses the carrier uses to source SIP and RTP traffic
  • 音声トラフィックには、G.711 コーデック(インバウンド/アウトバウンド)が推奨されます。

You can tell your SIP provider that these RFCs are supported by PureCloud.

標準 説明
RFC 791 Internet Protocol (IPv4)
RFC 2327 SDP: Session Description Protocol
RFC 2833

RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals

The customer endpoints should be able to handle in-band if RFC 2833 is not  supported (by the PBX and/or the far end).

RFC 3261

SIP: Session Initiation Protocol

The transport methods supported are UDP (RFC 768) and TCP (RFC 793).

The PBX-specific agreed transport method (i.e. UDP or TCP) shall apply in both  directions between the PBX and the service.

RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers
RFC 3264

An Offer/Answer Model with Session Description Protocol (SDP)

RFC 3311 The Session Initiation Protocol (SIP) UPDATE Method
RFC 3515 The Session Initiation Protocol (SIP) Refer Method
RFC 3550 RTP: A Transport Protocol for Real-Time Applications
RFC 3891 “Replaces” Header.
RFC 3892 Referred-By Mechanism.
RFC 3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP),
RFC 4028 Session Timers in the Session Initiation Protocol (SIP)
RFC 4244 An Extension to the Session Initiation Protocol (SIP) for Request History  Information
RFC 4566 SDP: Session Description Protocol (obsoletes RFC 2327)
RFC 4904 Representing Trunk Groups in tel/sip Uniform Resource Identifiers (URIs)
RFC 5806 Diversion Indication in SIP (network accepts does not send)
ITU G.168 Echo cancellation

Print the questionnaire before you contact your SIP provider. Then use it to discuss the SIP trunk configuration options with your SIP provider and record the answers. 

Download our SIP trunk questionnaire

You can then use the answers to configure each setting for the trunk configurations within PureCloud. For more information, see Create an external trunk.

Print the test plan and use it to perform the tests and record the results. If you have problems, you can then use this information to coordinate solutions with your SIP provider.

Download our SIP trunk test plan