本檔案為透過作業階段啟始通訊協定(SIP)/SIP思科整合邊界元件(CUBE)向各網際網路電話服務提供商(ITSP)的間歇性單向音訊傳出通話提供解決方案。
思科建議您瞭解SIP。
本文中的資訊係根據以下軟體和硬體版本:
思科整合通訊管理員(CUCM)
立方體
本文中的資訊是根據特定實驗室環境內的裝置所建立。文中使用到的所有裝置皆從已清除(預設)的組態來啟動。如果您的網路正在作用,請確保您已瞭解任何指令可能造成的影響。
如需文件慣例的詳細資訊,請參閱思科技術提示慣例。
通過SIP/SIP CUBE到各種ITSP的出站呼叫時間歇性單向音訊
呼叫流/拓撲:
建立者> CUCM(MGCP/SIP)> CUBE(SIP/SIP)> ITSP(Megafon)>終結者。
如果郵件傳輸代理(MTA)不支援會話描述協定(SDP)(REINVITE/200 OK)中的重複c=線路,則ITSP提供商會為從ITSP(Tx)到公共交換電話網路(PSTN)電話(Rx)的線路提供間歇性的單向音訊。
提供商:Megafon(Megacable)
沒有SIP配置檔案:
################################################################################ Sent: INVITE sip:3114560380@200.52.198.253:5151;transport=udp SIP/2.0 Via: SIP/2.0/UDP 200.52.198.15:5060;branch=z9hG4bK1BFE52263 From: <sip:3396900084@200.52.198.15:5060>;tag=3DF1D23A-15D3 To: sip:3114560380@200.52.198.253:5151;tag=227d2baf Date: Wed, 27 Feb 2013 19:44:31 GMT Call-ID: 00000196930006353732439410516722228326160@10.1.56.8 Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 360 Cisco-Guid: 3949497188-2152468962-2983459299-4054721625 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1361994271 Contact: <sip:3396900084@200.52.198.15:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 274 v=0 o=CiscoSystemsSIP-GW-UserAgent 8535 9331 IN IP4 200.52.198.15 s=SIP Call c=IN IP4 200.52.198.15 t=0 0 m=audio 18504 RTP/AVP 0 101 19 c=IN IP4 200.52.198.15 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=ptime:20
使用應用的SIP配置檔案:
注意: Connection-Info刪除第一個例項c=行,但不刪除第二個例項。
################################################################################ PSTN#show run | sec voice class sip-profile voice class sip-profiles 1000 request REINVITE sdp-header Connection-Info remove response 200 sdp-header Connection-Info remove Sent: INVITE sip:3310862061@200.52.198.253:5151;transport=udp SIP/2.0 Via: SIP/2.0/UDP 200.52.198.15:5060;branch=z9hG4bK1BFB91A7E From: <sip:3396900084@200.52.198.15:5060>;tag=3DC26466-1A5F To: MEGAFON <sip:3310862061@200.52.198.253:5151>;tag=3e3a03d7 Date: Wed, 27 Feb 2013 18:52:42 GMT Call-ID: 00000195730006353421530314263322228326160@10.1.56.8 Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 360 Cisco-Guid: 2932370470-2152010210-2968844771-4054721625 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 102 INVITE Max-Forwards: 70 Timestamp: 1361991162 Contact: <sip:3396900084@200.52.198.15:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 1274 9443 IN IP4 200.52.198.15 s=SIP Call t=0 0 m=audio 21846 RTP/AVP 0 101 19 c=IN IP4 200.52.198.15 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:19 CN/8000 a=ptime:20
使用應用的SIP配置檔案:
注意: Connection-Info將刪除第二個例項c=行,但不刪除第一個例項。
################################################################################ PSTN#show run | sec voice class sip-profile voice class sip-profiles 1000 request REINVITE sdp-header Audio-Connection-Info remove response 200 sdp-header Audio-Connection-Info remove Sent: INVITE sip:3310862061@200.52.198.253:5151;transport=udp SIP/2.0 Via: SIP/2.0/UDP 200.52.198.15:5060;branch=z9hG4bK1BFB91A7E From: <sip:3396900084@200.52.198.15:5060>;tag=3DC26466-1A5F To: MEGAFON <sip:3310862061@200.52.198.253:5151>;tag=3e3a03d7 Date: Wed, 27 Feb 2013 18:52:42 GMT Call-ID: 00000195730006353421530314263322228326160@10.1.56.8 Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 360 Cisco-Guid: 2932370470-2152010210-2968844771-4054721625 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 102 INVITE Max-Forwards: 70 Timestamp: 1361991162 Contact: <sip:3396900084@200.52.198.15:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 1274 9443 IN IP4 200.52.198.15 s=SIP Call c=IN IP4 200.52.198.15 t=0 0 m=audio 21846 RTP/AVP 0 101 19 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:19 CN/8000 a=ptime:20
*警告
SDP(RFC 2327)支援允許多個c線路,這顯示CUBE已正確實作該功能。此解決方案示例為不能正確支援RFC 2327的ITSP提供商提供了一個可能的解決方案。
在RFC中:
Session description v= (protocol version) o= (owner/creator and session identifier). s= (session name) i=* (session information) u=* (URI of description) e=* (email address) p=* (phone number) c=* (connection information - not required if included in all media) b=* (bandwidth information) One or more time descriptions (see below) z=* (time zone adjustments) k=* (encryption key) a=* (zero or more session attribute lines) Zero or more media descriptions (see below) Time description t= (time the session is active) r=* (zero or more repeat times) Media description m= (media name and transport address) i=* (media title) c=* (connection information - optional if included at session-level) b=* (bandwidth information) k=* (encryption key) a=* (zero or more media attribute lines)
使用此解決方案可解決此問題。
PSTN#show run | sec voice class sip-profile voice class sip-profiles 1000 request REINVITE sdp-header Audio-Connection-Info remove response 200 sdp-header Audio-Connection-Info remove
全域性設定配置檔案(語音服務VoIP)。
################################## PSTN#show run | sec voice service voip voice service voip sip sip-profiles 1000
設定特定撥號對等體上的配置檔案。這應在撥號對等體與PSTN之間設定。
################################################################### PSTN#show run | sec dial-peer voice 5566 dial-peer voice 5566 voip destination-pattern 6666 session target ipv4:1.1.1.1 voice-class sip profiles 1000
如需詳細資訊,請參閱使用SIP設定檔的Cisco整合邊界元件(CUBE)作業階段啟始通訊協定(SIP)規範化組態範例。
以下是支援的SDP標頭:
rtr(config-class)#response 200 sdp-header ? Attribute a= Audio-Attribute a= Audio-Bandwidth-Info b= Audio-Connection-Info c= Audio-Encryption-Key k= Audio-Media m=audio Audio-Session-Info I= Bandwidth-Key b= Connection-Info c= Email-Address e= Encrypt-Key k= Phone-Number p= Repeat-Times r= Session-Info I= Session-Name s= Session-Owner o= Time-Adjust-Key z= Time-Header t= Url-Descriptor u= Version v= Video-Attribute a= Video-Bandwidth-Info b= Video-Connection-Info c= Video-Encryption-Key k= Video-Media m=video Video-Session-Info I=
修訂 | 發佈日期 | 意見 |
---|---|---|
1.0 |
08-Mar-2013 |
初始版本 |