Solucao de Problemas No Session Initiation Protocol (SIP)

Webcast com Especialistas em
Tecnologia da Comunidade Cisco:
Soluo de Problemas envolvendo

Session Initiation Protocol (SIP)
Michelle Jardim
System Sales Engineer
Data: 06/12/2012
2011 Cisco and/or its affiliates. All rights reserved.
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O especialista de hoje a Engenheira de Sistema da Cisco
Michelle Jardim
Poder perguntar questes sobre Soluo de Problemas
envolvendo Session Initiation Protocol (SIP)
Foto do
Especialista
Michelle Jardim
CCIE Voice
34960
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A apresentao incluir algumas perguntas para o

pblico.
Convidamos voc a participar ativamente das

perguntas que faremos durante a sesso
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Se desejar baixar uma cpia da apresentao de hoje, v ao endereo

indicado no chat ou use este link
https://supportforums.cisco.com/docs/DOC-28412
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Qual o seu cargo em seu emprego atual?
a)
Administrador de redes
b)
Tcnico de informtica
c)
Analista de Sistemas
d)
Cargos gerenciais
e)
Outro
f)
Atualmente no trabalho
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Use o painel de perguntas e respostas (Q&A) para perguntar ao

especialista agora. Ele comear a responder.
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Michelle Jardim
CCIE Voice 34960
Data: 06/12/2012
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Viso geral do Session Initiation Protocol (SIP)

Ferramentas de Troubleshooting
Estudos de Caso
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Protocolo de sinalizao usado para estabelecer, gerenciar e
terminar sesses em uma rede IP

Protocolo definido na RFC 3261
Extendido em diversas outras RFCs
Baseado em mensagens ASCII

Equipamentos finais (Endpoints) so conhecidos como User
Agents
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Mensagens SIP
Pedidos e Respostas (Requests and Responses)
Cabealhos (Headers)
Negociao de Mdia
Session Description Protocol - SDP
Modelo Oferta/Resposta (Offer/Answer)
Early Offer x Delayed Offer
Early Media
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User Agent Clients (UAC) enviam Pedidos para os User Agent
Servers (UAS)
User Agent Servers enviam Respostas aos Pedidos
A maioria dos dispositivos SIP so ao mesmo tempo UAC e UAS
(ambos iniciam e aceitam Pedidos)

Unified CM e CUBE so Back-to-Back User Agents (B2BUA)
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INVITE - Um usurio ou servio est sendo convidado a
participar de uma sesso SIP

ACK - Confirma que um cliente recebeu uma Resposta final para
um Pedido de INVITE
BYE - Termina uma sesso existente; pode ser enviada por
qualquer User Agent (em uma sesso de mltiplos usurios)

CANCEL Cancela Pedidos pendentes; no finaliza sesses
que j foram aceitas

OPTIONS - Pergunta as capabilities de um servidor (Tambm
utilizado como Keep Alive)

REGISTER - Registra o User Agent no Registrar Server do
domnio
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INFO (RFC 2976) - para enviar mais informaes em um dilogo j estabelecido

PRACK (RFC 3262) - para confirmar o recebimento de uma Resposta
provisria a um Pedido.
SUBSCRIBE (RFC 3265) para informar que quer ser notificado se um evento
especfico ocorrer.
NOTIFY (RFC 3265) - para notificar um assinante que um evento especfico
ocorreu.
UPDATE (RFC 3311) - para atualizar parmetros antes de uma Resposta final
ao INVITE.
MESSAGE (RFC 3428) - Mensagens Instneas SIP
REFER (RFC 3515) - para remeter um User Agent para se comunicar com outro
User Agent
PUBLISH (RFC 3903) - para enviar informaes sobre status de um User Agent
a um Presence Server
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INVITE sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665
From: "Test User 1" <sip:[email protected]>;tag=97903bc0-43adcd-45510543
To: <sip:[email protected]>
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 2081204224-3137452793-0000000466-0996807340
Session-Expires: 1800
P-Asserted-Identity: "Test User 1" <sip:[email protected]>
Contact: <sip:[email protected]:5060>;video;audio
Max-Forwards: 69
Content-Length: 864
Content-Type: application/sdp
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CSeq: 101 INVITE
Expires: 180
Cisco-Guid: 2081204224-3137452793-0000000466-0996807340
Max-Forwards: 69
Content-Length: 864
URI
SIP Version
SIP Method
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CSeq: 101 INVITE
Expires: 180
Cisco-Guid: 2081204224-3137452793-0000000466-0996807340
Max-Forwards: 69
Content-Length: 864
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Response
Code
Description
Example
1xx
Provisional Request Received and

Continuing to Process Request
100 Trying
180 Ringing
183 Session Progress
2xx
Success Action was successfully received,

understood, and accepted
200 OK
202 Acceptable
3xx
Redirection Another SIP Element needs to

be contacted in order to complete the request
300 Multiple Choices

301 Moved Permanently
302 Moved Temporarily
4xx
Client Error Request contains bad syntax or

cannot be fulfilled at this server
401 Unauthorized
406 Not Acceptable
486 Busy Here
487 Request Terminated
488 Not Acceptable Here
5xx
Server Error Server failed to fulfill an

apparently valid request
503 Service Unavailable
6xx
Global Failure Request is invalid at any server
600 Busy Everywhere

603 Decline
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Unified CM
Phone 1
INVITE
200 OK
ACK
Session Established
BYE
200 OK
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Unified CM
Phone 1
SBC
(CUBE)
Phone
2
SBC
(CUBE)
CUBE
INVITE
INVITE
200 OK
200 OK
ACK
ACK
Session Established
BYE
BYE
200 OK
200 OK
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SP SBC
Unified CM
Phone 1
SBC (CUBE)
SIP SP
SBC
CUBE
INVITE
200 OK
ACK
INVITE
200 OK
INVITE
200 OK
ACK
ACK
Session Established
BYE
BYE
200 OK
BYE
200 OK
200 OK
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SIP utiliza o Session Description Protocol (SDP) (RFC
4566/3266/2327) para passar as informaes de mdia.

SIP usa o modelo Oferta/Resposta (Offer/Answer) descrito
na RFC 3264 para negociar a mdia utilizando o SDP.
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Um User Agent envia uma Oferta de SDP contendo toda as opes que
pode negociar.
SDP contm M linhas para cada media stream sendo negociado (Ex:
udio, vdeo, canal de contedo, etc)

O outro User Agent envia uma Resposta de SDP que contm as
mesmas ou parte das mesmas opes recebidas no pedido inicial.

De acordo com a RFC 3264, Para cada "m=" na Oferta, DEVE conter
uma linha correspondente "m=" na Resposta. A Resposta deve conter

o mesmo nmero de linhas "m=" contidas na Oferta.
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v=0
o=Cisco-SIPUA 26964 0 IN IP4 172.18.159.152
s=SIP Call
t=0 0
m=audio 29254 RTP/SAVP 0 8 18 102 9 116 124 101
c=IN IP4 172.18.159.152
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 25466 RTP/AVP 97
c=IN IP4 172.18.159.152
b=TIAS:1000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801E
a=recvonly
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v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 172.18.106.59
s=SIP Call
c=IN IP4 172.18.159.152
t=0 0
m=audio 30308 RTP/AVP 0 101
a=ptime:20
a=fmtp:101 0-15
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O User Agent Client pode enviar uma Oferta de SDP dentro
do INVITE - isto denominado Early Offer.

O User Agent Server pode enviar uma Oferta de SDP em
uma mensagem de Resposta caso o INVITE no contenha

uma Oferta de SDP - isto denominado Delayed Offer.
Para o Early Offer, a Resposta de SDP enviada dentro de
uma mensagem de Resposta ao INVITE.

Para o Delayed Offer, a Reposta de SDP enviada dentro
da mensagem de Pedido ACK.
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Unified CM
Phone 1
INVITE with SDP - Offer
200 OK with SDP - Answer

ACK (no SDP)
Session Established
BYE
200 OK
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Unified CM
Phone 1
INVITE (no SDP)
200 OK with SDP - Offer

ACK with SDP - Answer
Session Established
BYE
200 OK
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Delayed Offer no estabelece a mdia at a mensagem de
200 OK (ligao atendida)

Caso a mdia seja requerida antes da ligao ser atendida,
SIP executa o processo conhecido como Early Media

Com o Early Media em uma ligao Delayed Offer, a Oferta
de SDP originada pelo User Agent Server dentro de uma

mensagem de resposta Provisria (Ex: 183 Session
Progress)
O User Agent Client envia a Resposta de SDP dentro de
uma mensagem de pedido PRACK (Definida na RFC 3262)
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Unified CM
Phone 1
INVITE (no SDP)

183 Session Progress with SDP - Offer
PRACK with SDP - Answer
Media Stream Established

200 OK (PRACK)
200 OK (INVITE) w/ SDP (should be same as answer)
ACK
Session Established
BYE
200 OK
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Qualquer User Agent envolvido em uma ligao pode enviar um
Re-INVITE em um dilogo existente para renegociar parmetros

de uma ligao.
No possvel enviar um Re-INVITE antes que uma mensagem
de INVITE tenha recebido uma Resposta final.

O mtodo UPDATE tambm pode ser usado para renegociar,
porm antes que uma Resposta final tenha sido recebida.
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Re-INVITE Terminando uma Sesso de Mdia

INVITE sip:[email protected]:49833;transport=tls SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901f9c72c19221
From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6b82e2c213ca7-45545776
To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253be
Date: Wed, 11 Jan 2012 03:08:51 GMT
Min-SE: 1800
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 104 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= Authenticated; orientation= from; gci= 2-231448; callinstance= 2
Remote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5061;transport=tls>
Content-Length: 489
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Re-INVITE Terminando uma Sesso de Mdia

v=0
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 19594 RTP/SAVP 9 101
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
XXXXXXXXX
a=ptime:20
a=inactive
a=fmtp:101 0-15
b=TIAS:1000000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42801E;packetization-mode=1;level-asymmetry-allowed=1
a=inactive
a=mid:227796888
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Re-INVITE Delayed Offer para reestabelecer uma nova mdia

INVITE sip:[email protected]:49833;transport=tls SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1b
Date: Wed, 11 Jan 2012 03:08:52 GMT
Min-SE: 1800
CSeq: 106 INVITE
Max-Forwards: 70
Expires: 180
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= from; gci= 2-231448;
call-instance= 2
Remote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Content-Length: 0
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Re-INVITE Oferta na mensagem de resposta 200 OK

SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1b
From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca745545776
Date: Wed, 11 Jan 2012 03:08:52 GMT
CSeq: 106 INVITE
Server: Cisco-CPCIUS/9.2.1
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=called;idtype=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-ciscoescapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.2.0,Xcisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 788
Content-Disposition: session;handling=optional
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Re-INVITE Oferta na mensagem de Resposta 200 OK

v=0
s=SIP Call
t=0 0
m=audio 32518 RTP/SAVP 0 8 18 102 9 116 124 101
c=IN IP4 10.116.101.41
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=rtpmap:18 G729/8000
a=rtpmap:102 L16/16000
a=fmtp:101 0-15
a=sendrecv
m=video 17614 RTP/AVP 126 97
c=IN IP4 10.116.101.41
b=TIAS:2500000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42801F;packetization-mode=1;level-asymmetry-allowed=1
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F;packetization-mode=0;level-asymmetry-allowed=1
a=sendrecv
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Re-INVITE Resposta na mensagem de Pedido ACK

ACK sip:[email protected]:49833;transport=tls SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fb064465a06
From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d74931-4a07-83c6-b82e2c213ca7-45545776
Date: Wed, 11 Jan 2012 03:08:52 GMT
Max-Forwards: 70
CSeq: 106 ACK
Content-Length: 446
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Re-INVITE Resposta na mensagem de Pedido ACK
Rejeitando Suporte a vdeo

v=0
s=SIP Call
t=0 0
m=audio 4000 RTP/SAVP 0
c=IN IP4 172.18.106.58
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
XXXXXXXXXXXXXXXXXX
a=ptime:20
a=sendonly
c=IN IP4 10.116.101.50
b=TIAS:1000000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42801E;packetization-mode=1;level-asymmetry-allowed=1
a=mid:227796888
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Unified CM / SME Tools:

Real Time Monitoring Tool -> Session Trace
TranslatorX
Wireshark
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Permite procurar por uma chamada com base no nmero de origem ou
de destino
No depende de CDRs
Session trace apenas rastreia as sesses SIP, de forma detalhada
Exibe as mensagens SIP completas e originais

Utiliza tags de correlao para incluir todos os call legs relacionados
com a chamada selecionada

S pode ser usado para chamadas cujos traces ainda esto no servidor
(CUCM 9.0 permite visualizar traces que foram arquivados fora do

servidor)
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Diagrama do Fluxo da Chamada
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Passe o mouse para obter informaes sobre a mensagem
Clique em See SIP message para ver a mensagem SIP original
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Detalhes da Mensagem SIP
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Analisa arquivos de traces SDI do servio de CallManager

Decodifica mensagens SIP, SCCP, H.323, MGCP e ISDN Q.931
Lista de chamadas com base em mensagens de CDR nos Traces
Pode gerar diagramas de multi-protocolos
Capacidade avanada de filtragem

Download para Windows, Mac OS X e Linux em:
http://translatorx.cisco.com/
Guia sobre como habilitar e coletar via RTMT os traces do servio de
CallManager:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
a0080094e89.shtml
NOTA: O TAC no suporta a ferramenta TranslatorX
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Janela com Lista de Chamadas
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Filtrando a Lista de Chamadas
Select Each Call and click

Generate Filter button
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CDR
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CDR
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Filtros de Mensagens
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Ferramenta Open Source de captura e anlise de pacotes
Disponvel em http://www.wireshark.org
Disponvel para Windows, Mac OS X e UNIX / Linux
Oferece anlise de chamadas VoIP, incluindo SIP
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Anlise de Chamada VoIP
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Diagrama de Chamada VoIP
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Ambos, CUCM e IOS, fornecem mecanismo para coletar uma
captura de pacotes
Como realizar uma coleta de pacote?
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Acesse CUCM via SSH
Para comear a captura aplique o comando:
utils network capture file [file name] size all count 100000
Example:
utils network capture file CUCMCapture size all count 100000
Se este comando aplicado vrias vezes com o mesmo nome de
arquivo, CUCM ir automaticamente renomear o arquivo com _

e o seguinte nmero.
A captura parar depois que 100000 pacotes forem coletados ou
se CTRL+C pressionado.
Executar o teste recriando o problema.
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Faa download da captura usando um dos mtodos abaixo:
Se CUCM 7.X ou verses seguintes, colete os arquivos de log
via RTMT selecionando o servio Pacture Capture Logs (listado

na segunda pgina de servios) durante o perodo da captura.
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Se CUCM 6.X ou verses anteriores, aplique o comando
abaixo para enviar a captura para um servidor de SFTP:

file get activelog /platform/cli/[file name].cap
Example:
file get activelog /platform/cli/CUCMCapture.cap
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IP traffic Export feature permite capturar pacotes na interface de um roteador.

Configurao:
ip traffic-export profile CUBE_Debug mode capture
bidirectional
incoming access-list 101
outgoing access-list 101
interface GigabitEthernet0/0
ip traffic-export apply CUBE_Debug size 10000000
Utilizao:
traffic-export
traffic-export
traffic-export
traffic-export
interface g0/0 clear

interface g0/0 start
interface g0/0 stop
interface g0/0 copy tftp://10.1.1.1/capture.pcap
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Switched Port Analyzer feature permite capturar pacotes na interface de um
switch.
Configurao:
monitor session 1 source interface fastethernet 0/2
monitor session 1 destination interface fastethernet 0/3
Utilizao:
show monitor session 1
Guia para diferentes modelos de switch:
http://www.cisco.com/en/US/products/hw/switches/ps708/products_tech_note0918
6a008015c612.shtml#topic5
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Qual a sua experincia com as ferramentas

utilizadas para troubleshooting de SIP?
a)
Basica, J tive alguns contatos, porm no entendo muito

sobre a ferramenta.
b)
Eu tenho conhecimento avanado, porm frequentemente

utilizo os servios do TAC para auxiliar-me nesse processo.
c)
Estou em processo de aprendizado.
d)
Nenhuma.
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Descrio do Problema
Um usurio relata que toda vez que ele disca para (919) 555-
1212, recebe uma mensagem de que a chamada no pode

ser concluda da forma discada.
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Utilizando o RTMT Session Trace

Entre com *5551212 no campo Called Number/URI
Defina a hora de nicio e a durao da busca
A pesquisa encontrou duas chamadas
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Clique duas vezes para
ver o diagrama do fluxo

da chamada.
Mostra claramente que o
destino envia a
mensagem de Resposta
404 Not Found
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Troubleshoot da chamada no CUBE

Habilitar o debug de mensagens SIP debug ccsip messages
Jan 16 04:00:22.679: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Paul Giralt" <sip:[email protected]>;tag=19210128~0d0d25d7-4931-4a07-83c6-b82e2c213ca745568313
Date: Mon, 16 Jan 2012 03:55:17 GMT
Min-SE: 1800
CSeq: 101 INVITE
Expires: 180
Cisco-Guid: 3852191232-0000065536-0000018595-0996807340
P-Asserted-Identity: "Paul Giralt" <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Max-Forwards: 69
Content-Length: 0
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Troubleshoot da chamada no CUBE

Jan 16 04:00:22.687: //98/E59BC6000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
To: <sip:[email protected]>;tag=253488-726
Date: Mon, 16 Jan 2012 04:00:22 GMT
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.2.T
Reason: Q.850;cause=1
Content-Length: 0
Verifique se o nmero corresponde a um dial peer vlido:

show dialplan number +19195551212
Macro Exp.: +19195551212
No match, result=-1
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Descrio do Problema
Um usurio relata que toda vez que ele disca para (919) 555-
1212, ele recebe um reorder (fast busy) tone.
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Entre com *5551212 no campo Called Number/URI
Defina a hora de nicio e a durao da busca
A pesquisa encontrou duas chamadas
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O diagrama
mostra a
sinalizao a
partir do
telefone, e
tambm para o
CUBE
O CUBE est
respondendo
com uma
mensagem de
Resposta 403
Forbidden
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A partir da verso 15.1(2) T, o IOS rejeita chamadas de origens
desconhecidas por padro.

possvel desativar essa feature ou adicionar lista de endereos
permitidos:
voice service voip
no ip address trusted authenticate
allow-connections sip to sip
sip
voice service voip
ip address trusted list
ipv4 172.18.106.0 255.255.255.0
allow-connections sip to sip
sip
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Descrio do Problema
Um usurio relata que toda vez que liga para um nmero de
telefone especfico, ningum atende a chamada; mas se ele

chama esse mesmo nmero a partir de seu telefone celular, a
chamada sempre atendida.
Nmero chamador 89919236.
Nmero chamado 1 877 288-8362.
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Coletando os Traces
Como possvel recriar o problema, pode gerar uma chamada de
teste e depois recolher os traces do CallManager.
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Utilizando TranslatorX para anlise dos traces

Select File > Open Folder
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Utilizando o TranslatorX para anlise dos Traces

Pesquisar pelo nmero chamado
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Desabilite os filtros
Selecione o INVITE
Filtre pelo SIP Call ID
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03/29/2010 10:36:41.497 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to
172.18.159.231:[5060]:
From: "Test User 1" <sip:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd45510543
Date: Mon, 29 Mar 2010 14:36:41 GMT
Min-SE: 1800
CSeq: 101 INVITE
Expires: 180
Cisco-Guid: 2081204224-3137452793-0000000466-0996807340
Max-Forwards: 69
Content-Length: 0
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Onde que a chamada foi originada? Tente pesquisar pelo nmero do
chamador
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Selecione o INVITE
Crie novo filtro
Filtre pelo Endereo IP
Habilite os filtros
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Cisco Confidential
85
INVITE do IP Phone com SDP ao CUCM

03/29/2010 10:36:33.771 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on
port 51682 index 2321 with 1717 bytes:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61
From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0c
To: <sip:[email protected];user=phone>
Max-Forwards: 70
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP9951/9.0.1
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Test User 1" <sip:[email protected]>;party=calling;idtype=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-ciscoescapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.0.0,Xcisco-xsi-9.0.1
Allow-Events: kpml,dialog
Content-Length: 632
Content-Disposition: session;handling=optional
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INVITE do IP Phone com SDP ao CUCM (continuao)

v=0
s=SIP Call
t=0 0
m=audio 29254 RTP/SAVP 0 8 18 102 9 116 124 101
c=IN IP4 172.18.159.152
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=fmtp:116 mode=20
a=fmtp:101 0-15
a=sendrecv
c=IN IP4 172.18.159.152
b=TIAS:1000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801E
a=recvonly
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Unified CM envia 100 Trying

03/29/2010 10:36:33.773 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on
port 51682 index 2321
SIP/2.0 100 Trying
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 INVITE
Content-Length: 0
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Unified CM envia REFER para tocar Outside Dialtone

03/29/2010 10:36:33.780 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port
51682 index 2321
REFER sip:[email protected]:51682 SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151511c5f04bf
From: <sip:[email protected]>;tag=2144536187
CSeq: 101 REFER
Max-Forwards: 70
Expires: 0
Refer-To: cid:[email protected]
Content-Id: <[email protected]>
Require: norefersub
Content-Type: application/x-cisco-remotecc-request+xml
Referred-By: <sip:[email protected]>
Content-Length: 409
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Unified CM envia REFER para tocar Outside Dialtone (continuao)

<x-cisco-remotecc-request>
<playtonereq>
<dialogid>
<callid>[email protected]</callid>
<localtag>97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542</localtag>
<remotetag>00260bd9669e07147bcb3aac-3cda8f0c</remotetag>
</dialogid>
<tonetype>DtOutsideDialTone</tonetype>
<direction>user</direction>
</playtonereq>
</x-cisco-remotecc-request>
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Unified CM envia um SUBSCRIBE for KPML

03/29/2010 10:36:33.781 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port
51682 index 2321
SUBSCRIBE sip:[email protected]:51682 SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK1515232b4e84f
CSeq: 101 SUBSCRIBE
Date: Mon, 29 Mar 2010 14:36:33 GMT
Event: kpml; [email protected]; fromtag=00260bd9669e07147bcb3aac-3cda8f0c
Expires: 7200
Accept: application/kpml-response+xml
Max-Forwards: 70
Content-Type: application/kpml-request+xml
Content-Length: 424
<?xml version="1.0" encoding="UTF-8" ?>
<kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchemainstance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0">
<pattern criticaldigittimer="1000" extradigittimer="500" interdigittimer="10000" persist="persist">
<regex tag="Backspace OK">[x#*+]|bs</regex>
</pattern>
</kpml-request>
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Phone envia 200 OK para REFER e SUBSCRIBE

03/29/2010 10:36:33.802 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682
index 2321 with 453 bytes:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151511c5f04bf
To: <sip:[email protected]>;tag=00260bd9669e07167c743311-343ee3af
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 REFER
Server: Cisco-CP9951/9.0.1
Contact: <sip:[email protected]:51682;transport=TLS>
Content-Length: 0
03/29/2010 10:36:33.843 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682
index 2321 with 465 bytes:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK1515232b4e84f
To: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 SUBSCRIBE
Expires: 7200
Content-Length: 0
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IP Phone
Unified CM
SIP Gateway
(172.18.159.152)
(172.18.106.59)
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
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Usurio disca o 1
NOTIFY sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1cd529ba
To: <sip:[email protected]>;tag=1976165806
From: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 1001 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK"
suppressed="false" forced_flush="false" digits="1" tag="Backspace OK"/>
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Unified CM responde NOTIFY With a 200 OK

SIP/2.0 200 OK
Date: Mon, 29 Mar 2010 14:36:34 GMT
CSeq: 1001 NOTIFY
Content-Length: 0
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Unified CM envia REFER para parar Outside Dialtone

REFER sip:[email protected]:51682 SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151536ea86ab0
CSeq: 101 REFER
Max-Forwards: 70
Expires: 0
Refer-To: cid:[email protected]
Content-Id: <[email protected]>
Require: norefersub
Content-Type: application/x-cisco-remotecc-request+xml
Referred-By: <sip:[email protected]>
Content-Length: 401
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Unified CM envia REFER para parar Outside Dialtone (Continuao)

<x-cisco-remotecc-request>
<playtonereq>
<dialogid>
<callid>[email protected]</callid>
<localtag>97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542</localtag>
<remotetag>00260bd9669e07147bcb3aac-3cda8f0c</remotetag>
</dialogid>
<tonetype>Dt_NoTone</tonetype>
<direction>user</direction>
</playtonereq>
</x-cisco-remotecc-request>
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Telefone responde com 200 OK para REFER

SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151536ea86ab0
To: <sip:[email protected]>;tag=00260bd9669e07184b08b96b-796ab86f
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 REFER
Content-Length: 0
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IP Phone
Unified CM
SIP Gateway
(172.18.159.152)
(172.18.106.59)
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
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Usrio disca o 8
03/29/2010 10:36:34.944 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from
172.18.159.152 on port 51682 index 2321 with 896 bytes:
NOTIFY sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK647d03c1
Date: Mon, 29 Mar 2010 14:36:34 GMT
CSeq: 1002 NOTIFY
Event: kpml
Subscription-State: active; expires=7195
Max-Forwards: 70
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200"
text="OK" suppressed="false" forced_flush="false" digits="8" tag="Backspace OK"/>
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Unified CM responde NOTIFY With a 200 OK

03/29/2010 10:36:34.352 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to
172.18.159.152 on port 51682 index 2321
SIP/2.0 200 OK
Date: Mon, 29 Mar 2010 14:36:34 GMT
CSeq: 1001 NOTIFY
Content-Length: 0
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Usurio disca o restantes dos dgitos
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IP Phone
Unified CM
SIP Gateway
(172.18.159.152)
(172.18.106.59)
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
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Digit Analysis
10:36:41.486 |Digit analysis: match(pi="2", fqcn="+19194769236", cn="89919236",plv="5",
pss="1stLine:RTP_AbbrDial:Cisco:US Local:US RTP Local:US Long Distance:US
International:VMPilotPartition", TodFilteredPss="1stLine:RTP_AbbrDial:Cisco:US Local:US RTP Local:US Long
Distance:US International:VMPilotPartition", dd="918772888362",dac="1)
10:36:41.486 |Digit analysis: analysis results
10:36:41.486 ||PretransformCallingPartyNumber=+19194769236
|CallingPartyNumber=+19194769236
|DialingPartition=GDP_GlobalE164_PSTN
|DialingPattern=\+1.[2-9]XX[2-9]XXXXXX
|FullyQualifiedCalledPartyNumber=+18772888362
|DialingPatternRegularExpression=(+1)([2-9][0-9][0-9][2-9][0-9][0-9][0-9][0-9][0-9][0-9])
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=+18772888362
|PretransformTagsList=ACCESS-CODE:SUBSCRIBER
|PretransformPositionalMatchList=+1:8772888362
|CollectedDigits=+18772888362
|UnconsumedDigits=
|TagsList=ACCESS-CODE:SUBSCRIBER
|PositionalMatchList=+1:8772888362
|VoiceMailbox=
|VoiceMailCallingSearchSpace=1stLine:RTP_AbbrDial
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Digit Analysis (Continuao)

|VoiceMailPilotNumber=89944444
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=
|UnicodeDisplayName=
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
|WithTags=
|WithValues=
|CallingPartyNumberPi=NotSelected
|ConnectedPartyNumberPi=NotSelected
|CallingPartyNamePi=NotSelected
|ConnectedPartyNamePi=NotSelected
|CallManagerDeviceType=NoDeviceType
|PatternPrecedenceLevel=Routine
|CallableEndPointName=[23146446-6606-7227-3882-75d07dd6fdef]
|PatternNodeId=[9badd465-d20a-5bc7-1077-8edee47e8caf]
|AARNeighborhood=[]
|AARDestinationMask=[]
|AARKeepCallHistory=true
|AARVoiceMailEnabled=false
|NetworkLocation=OffNet
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|Calling Party Number Type=Cisco Unified CallManager
|Calling Party Numbering Plan=Cisco Unified CallManager
|Called Party Number Type=Cisco Unified CallManager
|Called Party Numbering Plan=Cisco Unified CallManager
|ProvideOutsideDialtone=false
|AllowDeviceOverride=false
|AlternateMatches=
{
|Partition=US Long Distance
{
<
|Pattern=9.1[2-9]XX[2-9]XXXXXX
|PatternType=Translation
|TranslationPartition=[a6bd708e-ac4d-ae55-3134-b90b987e5ad9]
|PatternPrecedenceLevel=PlDefault
|PatternRouteClass=RouteClassDefault
|RouteNextHopByCgpn=false
>
}
}
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|TranslationPatternDetails=
|PretransformCallingPartyNumber=89919236
|CallingPartyNumber=+19194769236
|DialingPartition=US Local
|DialingPattern=9.1877[2-9]XXXXXX
|FullyQualifiedCalledPartyNumber=918772888362
|DialingPatternRegularExpression=(9)(1877[2-9][0-9][0-9][0-9][0-9][0-9][0-9])
|DialingWhere=
|PatternType=Translation
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=918772888362
|PretransformTagsList=ACCESS-CODE:SUBSCRIBER
|PretransformPositionalMatchList=9:18772888362
|CollectedDigits=+18772888362
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=18772888362
|VoiceMailbox=
|VoiceMailCallingSearchSpace=
|VoiceMailPilotNumber=
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=1
|AlertingName=
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|UnicodeDisplayName=
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
|WithTags=
|WithValues=
|CallingPartyNumberPi=NotSelected
|ConnectedPartyNumberPi=NotSelected
|CallingPartyNamePi=NotSelected
|ConnectedPartyNamePi=NotSelected
|PatternPrecedenceLevel=Routine
|CallableEndPointName=[bb6f140a-5fd4-179a-2cad-2a1d5eacca7e]
|PatternNodeId=[bb6f140a-5fd4-179a-2cad-2a1d5eacca7e]
|AARNeighborhood=[]
|AARDestinationMask=[]
|AARKeepCallHistory=true
|AARVoiceMailEnabled=false
|NetworkLocation=OnNet
|Calling Party Number Type=Cisco Unified CallManager
|Calling Party Numbering Plan=Cisco Unified CallManager
|Called Party Number Type=Cisco Unified CallManager
|Called Party Numbering Plan=Cisco Unified CallManager
|ProvideOutsideDialtone=true
|AllowDeviceOverride=false
|AlternateMatches=
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CUCM envia INVITE para o PSTN Gateway

03/29/2010 10:36:41.497 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.18.159.231:[5060]:
Date: Mon, 29 Mar 2010 14:36:41 GMT
Min-SE: 1800
CSeq: 101 INVITE
Expires: 180
Cisco-Guid: 2081204224-3137452793-0000000466-0996807340
Max-Forwards: 69
Content-Length: 0
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Unified CM
IP Phone
SIP Gateway
(172.18.106.59)
(172.18.159.152)
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
200 OK (SUBSCRIBE)
INVITE
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Gateway responde com 183 Session Progress com SDP

03/29/2010 10:36:42.324 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 1568 from
172.18.159.231:[5060]:
SIP/2.0 183 Session Progress
To: <sip:[email protected]>;tag=DE1EFF8-0
Date: Mon, 29 Mar 2010 14:37:23 GMT
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO,
REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 788
--uniqueBoundary
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Gateway responde 183 Session Progress com SDP (Continuao)

v=0
o=CiscoSystemsSIP-GW-UserAgent 0 7954 IN IP4 172.18.159.231
s=SIP Call
c=IN IP4 172.18.159.231
t=0 0
m=audio 27980 RTP/AVP 0 8 116 18 100 101
c=IN IP4 172.18.159.231
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=fmtp:101 0-16
--uniqueBoundary
Content-Type: application/x-q931
Content-Disposition: signal;handling=optional
Content-Length: 11
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Unified CM envia 180 Ringing ao IP Phone

SIP/2.0 180 Ringing
To: <sip:[email protected];user=phone>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 INVITE
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= to; ui-state= ringout; gci= 2305505; call-instance= 1
Send-Info: conference
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Content-Length: 0
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IP Phone
Unified CM
SIP Gateway
(172.18.159.152)
(172.18.106.59)
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
200 OK (SUBSCRIBE)
INVITE
100 Trying
183 Session Progress
180 Ringing
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Telefone continua tocando

Timestamps salta de 10:36:42 para 10:37:32
Sem sinalizao SIP por 50 segundos
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Telefone envia uma CANCEL

CANCEL sip:[email protected];user=phone SIP/2.0
Max-Forwards: 70
Date: Mon, 29 Mar 2010 14:37:32 GMT
CSeq: 101 CANCEL
User-Agent: Cisco-CP9951/9.0.1
Content-Length: 0
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Unified CM envia um 200 OK pelo CANCEL

SIP/2.0 200 OK
Date: Mon, 29 Mar 2010 14:37:32 GMT
CSeq: 101 CANCEL
Content-Length: 0
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Unified CM
IP Phone
SIP Gateway
(172.18.106.59)
(172.18.159.152)
(172.18.159.231)
NOTIFY
200 OK (NOTIFY)
CANCEL
200 OK (CANCEL)
CANCEL
487 Request Cancelled
200 OK (CANCEL)
487 Request Cancelled
ACK
ACK
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O que fazer para que o gateway abra canal de voz aps a mensagem
183 Session Progress?

RFC 3262: Reliability of Provisional Responses in the Session Initiation
Protocol (SIP)
Fornece uma forma de reconhecer a mensagem 183 Session Progress
PRACK
Parmetro SEP Rel1XX Options * no CUCM:
Disabled
Send PRACK for all 1xx Messages
Send PRACK if 1xx Contains SDP
*Service Parameter in 7.x and earlier. SIP Profile parameter in 8.x and later
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Qual dos comandos abaixo, habilita o debug

de mensagens SIP?
a)
Debug voice ccapi inout
b)
Debug ccsip messages
c)
Debug voip vtsp default
d)
Debug voip vtsp session
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121
O Especialista responder s questes verbalmente. Use o painel

escrito Q&A para continuar fazendo suas perguntas.
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122
Aqueles que preencherem

o questionrio de avaliao
entraro em um sorteio para
ganhar:
Um vale presente
Para fazer a avaliao, favor clicar no endereo

fornecido no chat ou no pop-up quando o evento
terminar.
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123
Se tiver perguntas adicionais, poder perguntar ao especialista

Ele estar respondendo do dia 06 de Dezembro ao dia 17 de
Dezembro.
https://supportforums.cisco.com/thread/2186372
Voc poder assistir ao video ou ler as perguntas e respostas 5 dias
teis aps o evento em
https://supportforums.cisco.com/community/portuguese/canto-dosespecialistas/webcasts
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124
Tema: Melhores prticas para migrao do ASA

(Adaptive Security Appliance) para Release 8.3/8.4
Com o Especialista Cisco : Paulo Aguiar
Tire dvidas sobre solues da Cisco em Melhores
prticas para migrao do ASA (Adaptive Security
Appliance) para Release 8.3/8.4
Este evento estar aberto at o dia 07 de Dezembro
Acesse
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Solucao de Problemas No Session Initiation Protocol (SIP)

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Solucao de Problemas No Session Initiation Protocol (SIP)

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Webcast com Especialistas em

Tecnologia da Comunidade Cisco:

Soluo de Problemas envolvendo

2011 Cisco and/or its affiliates. All rights reserved.

O especialista de hoje a Engenheira de Sistema da Cisco

envolvendo Session Initiation Protocol (SIP)

A apresentao incluir algumas perguntas para o

Convidamos voc a participar ativamente das

2011 Cisco and/or its affiliates. All rights reserved.

Se desejar baixar uma cpia da apresentao de hoje, v ao endereo

2011 Cisco and/or its affiliates. All rights reserved.

Qual o seu cargo em seu emprego atual?

2011 Cisco and/or its affiliates. All rights reserved.

Use o painel de perguntas e respostas (Q&A) para perguntar ao

2011 Cisco and/or its affiliates. All rights reserved.

2011 Cisco and/or its affiliates. All rights reserved.

Viso geral do Session Initiation Protocol (SIP)

2011 Cisco and/or its affiliates. All rights reserved.

2011 Cisco and/or its affiliates. All rights reserved.

Protocolo de sinalizao usado para estabelecer, gerenciar e

terminar sesses em uma rede IP

Baseado em mensagens ASCII

2011 Cisco and/or its affiliates. All rights reserved.

2011 Cisco and/or its affiliates. All rights reserved.

User Agent Clients (UAC) enviam Pedidos para os User Agent

(ambos iniciam e aceitam Pedidos)

2011 Cisco and/or its affiliates. All rights reserved.

INVITE - Um usurio ou servio est sendo convidado a

participar de uma sesso SIP

qualquer User Agent (em uma sesso de mltiplos usurios)

que j foram aceitas

utilizado como Keep Alive)

INFO (RFC 2976) - para enviar mais informaes em um dilogo j estabelecido

INVITE sip:[email protected]:5060 SIP/2.0

INVITE sip:[email protected]:5060 SIP/2.0

2011 Cisco and/or its affiliates. All rights reserved.

INVITE sip:[email protected]:5060 SIP/2.0

Provisional Request Received and

Success Action was successfully received,

Redirection Another SIP Element needs to

300 Multiple Choices

Client Error Request contains bad syntax or

Server Error Server failed to fulfill an

503 Service Unavailable

Global Failure Request is invalid at any server

600 Busy Everywhere

2011 Cisco and/or its affiliates. All rights reserved.

2011 Cisco and/or its affiliates. All rights reserved.

2011 Cisco and/or its affiliates. All rights reserved.

2011 Cisco and/or its affiliates. All rights reserved.

SIP utiliza o Session Description Protocol (SDP) (RFC

4566/3266/2327) para passar as informaes de mdia.

na RFC 3264 para negociar a mdia utilizando o SDP.

2011 Cisco and/or its affiliates. All rights reserved.

udio, vdeo, canal de contedo, etc)

mesmas ou parte das mesmas opes recebidas no pedido inicial.

uma linha correspondente "m=" na Resposta. A Resposta deve conter

2011 Cisco and/or its affiliates. All rights reserved.

2011 Cisco and/or its affiliates. All rights reserved.

O User Agent Client pode enviar uma Oferta de SDP dentro