Este documento fornece uma solução para chamadas de saída de áudio unidirecionais intermitentes pelo Session Initiation Protocol (SIP)/SIP Cisco Unified Border Element (CUBE) para vários Provedores de Serviços de Telefonia pela Internet (ITSPs).
A Cisco recomenda que você tenha conhecimento do SIP.
As informações neste documento são baseadas nestas versões de software e hardware:
Cisco Unified Communications Manager (CUCM)
CUBO
As informações neste documento foram criadas a partir de dispositivos em um ambiente de laboratório específico. Todos os dispositivos utilizados neste documento foram iniciados com uma configuração (padrão) inicial. Se a sua rede estiver ativa, certifique-se de que entende o impacto potencial de qualquer comando.
Consulte as Convenções de Dicas Técnicas da Cisco para obter mais informações sobre convenções de documentos.
Áudio unidirecional intermitente em chamadas de saída pelo CUBE SIP/SIP para vários ITSPs
Fluxo de chamada/topologia:
Originador > CUCM (MGCP/SIP) > CUBE (SIP/SIP) > ITSP (Megafon) > Terminador.
Os provedores de ITSP que têm Mail Transfer Agents (MTA) que não suportam linhas c= duplicadas no Session Description Protocol (SDP) (REINVITE/200 OK) causam áudio unidirecional intermitente para o trecho do ITSP(Tx) para o telefone (Rx) da rede telefônica pública comutada (PSTN).
Provedor(es): Megafon (Megacable)
Sem perfil SIP:
################################################################################ 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
Com perfil SIP aplicado:
Observação: Connection-Info remove as linhas da primeira instância c=, mas não a segunda.
################################################################################ 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
Com perfil SIP aplicado:
Observação: Connection-Info remove as linhas da segunda instância c=, mas não a primeira.
################################################################################ 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
*Aviso
O suporte a SDP (RFC 2327) permite várias linhas c, o que mostra que o CUBE implementou corretamente o recurso. Este exemplo de solução serve como uma solução possível para provedores de ITSP que não suportam corretamente o RFC 2327.
Do RFC:
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)
Use esta solução para resolver o problema.
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
Defina o perfil globalmente (VoIP de serviço de voz).
################################## PSTN#show run | sec voice service voip voice service voip sip sip-profiles 1000
Defina o perfil em um peer de discagem específico. Isso deve ser definido no peer de discagem de e para o PSTN.
################################################################### 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
Consulte o documento Normalização do Session Initiation Protocol (SIP) do Cisco Unified Border Element (CUBE) com Exemplo de Configuração de Perfis SIP para obter mais informações.
Estes são os cabeçalhos SDP suportados:
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=
Revisão | Data de publicação | Comentários |
---|---|---|
1.0 |
08-Mar-2013 |
Versão inicial |