Dépannage d'un appel SIP entre deux terminaux

Introduction

Ce document fournit un exemple de configuration de deux télécopieurs afin d’expliquer comment un appel de protocole SIP (Session Initiation Protocol) a lieu entre deux passerelles. Ce document fournit également une explication sur la sortie de la commande debug ccsip messages pour le dépannage des échecs d’appel SIP.

Conditions préalables

Conditions requises

Aucune spécification déterminée n'est requise pour ce document.

Components Used

Les informations contenues dans ce document sont basées sur les versions de matériel et de logiciel suivantes :

• Deux télécopieurs

• VG224 qui exécute le logiciel Cisco IOS^® Version 12.4(4)T1

• Routeur Cisco 3745 qui exécute le logiciel Cisco IOS Version 12.3(11)T8

The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, make sure that you understand the potential impact of any command.

Conventions

Pour plus d'informations sur les conventions utilisées dans ce document, reportez-vous à Conventions relatives aux conseils techniques Cisco.

Configuration

Cette section vous fournit des informations pour configurer les fonctionnalités décrites dans ce document.

Remarque : Utilisez l’outil de recherche de commandes (clients inscrits seulement) pour en savoir plus sur les commandes figurant dans le présent document.

Diagramme du réseau

Ce document utilise la configuration réseau suivante :

Configurations

Ce document utilise les configurations suivantes :

• VG224

• Cisco 3745

vg224#show run
Building configuration...
!
voice call send-alert
voice rtp send-recv
!
voice service pots
!
voice service voip
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0
!
voice-port 2/0
 idle-voltage low
!
dial-peer voice 1 pots
<fax machine connected to this port>
 destination-pattern 9000
 port 2/0
!
dial-peer voice 100 voip
 destination-pattern 8000
 no modem passthrough
 session protocol sipv2
 session target ipv4:172.16.184.83
 incoming called-number .
 codec g711ulaw
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
!

HTTS-VRK1-3745-1#show run
Building configuration...
!
voice service voip
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0
!
!
voice-port 4/1/0
!
!
dial-peer voice 9000 voip
 destination-pattern 9000
 session protocol sipv2
 session target ipv4:172.16.13.87
 incoming called-number .
 codec g711ulaw
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
 no vad
!
dial-peer voice 9 pots
 destination-pattern 8000
 fax rate voice
 port 4/1/0
 forward-digits all

Vérification

Aucune procédure de vérification n'est disponible pour cette configuration.

Dépannage

Utilisez cette section pour dépanner votre configuration.

L'Outil Interpréteur de sortie (clients enregistrés uniquement) (OIT) prend en charge certaines commandes show. Utilisez l'OIT pour afficher une analyse de la sortie de la commande show .

Remarque : Consulter les renseignements importants sur les commandes de débogage avant d’utiliser les commandes de débogage.

Voici la sortie de la commande debug ccsip messages :

!--- This is the first invite message sent out !--- to the terminating SIP gateway. !--- This is similar to a setup message in H.323 or Q.931.

*Mar 1 00:33:42.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:8000@172.16.184.83:5060 SIP/2.0

!--- 8000 is the DN of the call, 172.16.184.83 is !--- the IP address of the remote gateway, and !--- 5060 is the port the SIP works on. !--- This configuration uses SIP version 2.0.

Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKB21AF

!--- The VIA field is used for devices in the patch that !--- need to be aware of the call. !--- In this case, there are no SIP devices in between the two gateways.

Remote-Party-ID: <sip:9000@172.16.13.87>;party=calling;screen=no;privacy=off
 !--- The DN and URI of the remote SIP device that is called.

From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>
Date: Fri, 01 Mar 2002 00:33:42 GMT

!--- The time that the invite is sent out

Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87

!--- The call ID is unique for every call. !--- This ID is used to identify a particular call !--- in a busy router.

Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 3481906499-736235990-2149183265-3714191467
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE

!--- The sequence number for each transaction.
 
Max-Forwards: 70
Timestamp: 1014942822
Contact: <sip:9000@172.16.13.87:5060>

!--- This is the address used to reach the calling party on the return path.

Expires: 180

!--- This message expires in 180 seconds.

Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 215

v=0
 !--- The Session Descriptor Protocol (SDP) version is zero. !--- This is different from the SIP version used !--- in this example configuration.

o=CiscoSystemsSIP-GW-UserAgent 1715 2724 IN IP4 172.16.13.87

!--- The owner of the device that created the call. !--- This is sometimes referred to as organization.

s=SIP Call

!--- The name given to this particular SIP call. This is the description.

c=IN IP4 172.16.13.87

!--- Connection information. Usually includes the IP address of !--- the originating device. It is an optional field.

t=0 0
m=audio 18080 RTP/AVP 0 19

!--- This is the media information. In this case, !--- 18080 is used as the UDP port for RTP.

c=IN IP4 172.16.13.87
a=rtpmap:0 PCMU/8000

!--- This is the media attributes. Notice the 0 and 19 in !--- the media field. These are the !--- attributes that go with that. PCMU/8000 is G711ulaw.

a=rtpmap:19 CN/8000
a=ptime:20

!--- A packetization period of 20 ms.



!--- In this output, invite, SDP is not a required parameter. !--- But in this case you see that SDP sent out. !--- SDP carries information about capabilities. !--- No information about fax capabilities are !--- exchanged in the beginning because it is only a voice !--- call until you hear fax tones from the terminating fax machine.


*Mar 1 00:33:43.203: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKB21AF
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Tue, 28 Feb 2006 23:43:36 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Timestamp: 1014942822
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow-Events: telephone-event
Content-Length: 0


!--- The terminating machine sets up an analog !--- connection to the fax machine, and while it waits, !--- it sends a "trying" message. This stops the !--- originating gateway from sending another invite.


*Mar 1 00:33:43.207: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKB21AF
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Tue, 28 Feb 2006 23:43:36 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Timestamp: 1014942822
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Require: 100rel
RSeq: 3696
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:8000@172.16.184.83:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 7643 2735 IN IP4 172.16.184.83
s=SIP Call
c=IN IP4 172.16.184.83
t=0 0
m=audio 18304 RTP/AVP 0

!--- This is a different UDP port for the reverse direction.

c=IN IP4 172.16.184.83
a=rtpmap:0 PCMU/8000
a=ptime:20


!--- A "progress" indicator tells you that the remote gateway sent a connect !--- and the fax machine is ringing at this time. !--- Note that the To and From headers do not change despite !--- the fact that the message comes in the opposite direction.


*Mar 1 00:33:43.211: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKB21AF
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Tue, 28 Feb 2006 23:43:36 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Timestamp: 1014942822
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Require: 100rel
RSeq: 3696
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:8000@172.16.184.83:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 7643 2735 IN IP4 172.16.184.83
s=SIP Call
c=IN IP4 172.16.184.83
t=0 0
m=audio 18304 RTP/AVP 0
c=IN IP4 172.16.184.83
a=rtpmap:0 PCMU/8000
a=ptime:20


!--- A provisional ack to the progress message.


*Mar 1 00:33:43.251: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
PRACK sip:8000@172.16.184.83:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKC384
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Fri, 01 Mar 2002 00:33:42 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
CSeq: 102 PRACK
RAck: 3696 101 INVITE
Max-Forwards: 70
Content-Length: 0


!--- This is an OK for the PRACK. You can tell this from the Cseq header.


*Mar 1 00:33:44.031: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKC384
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Tue, 28 Feb 2006 23:43:37 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 PRACK
Content-Length: 0


!--- An OK is received, which is mandatory for an invite. !--- The OK has information on the accepted media parameters in the SDP.


*Mar 1 00:33:49.431: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKB21AF
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Tue, 28 Feb 2006 23:43:37 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Timestamp: 1014942822
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:8000@172.16.184.83:5060>
Content-Type: application/sdp
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 7643 2735 IN IP4 172.16.184.83
s=SIP Call
c=IN IP4 172.16.184.83
t=0 0
m=audio 18304 RTP/AVP 0
c=IN IP4 172.16.184.83
a=rtpmap:0 PCMU/8000
a=ptime:20


!--- The ack for the OK.


*Mar 1 00:33:49.443: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
ACK sip:8000@172.16.184.83:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.13.87:5060;branch=z9hG4bKD1A5C
From: <sip:9000@172.16.13.87>;tag=1EDC10-2436
To: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
Date: Fri, 01 Mar 2002 00:33:42 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0


!--- At this point, the terminating gateway hears fax tones and determines it !--- has to switch the codec to a !--- fax codec and sends a re-invite. The re-invite contains !--- information about the new media !--- parameters that the terminating gateway wants to change to. 

*Mar 1 00:33:55.247: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
INVITE sip:9000@172.16.13.87:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1A735
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Tue, 28 Feb 2006 23:43:49 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Supported: 100rel,timer
Min-SE: 1800
Cisco-Guid: 3481906499-736235990-2149183265-3714191467
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, 
NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1141170229
Contact: <sip:8000@172.16.184.83:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 399

v=0
o=CiscoSystemsSIP-GW-UserAgent 7643 2736 IN IP4 172.16.184.83
s=SIP Call
c=IN IP4 172.16.184.83
t=0 0
m=image 18304 udptl t38
c=IN IP4 172.16.184.83
a=T38FaxVersion:0
a=T38MaxBitRate:14400

!--- The maximum bit rate that is supported by the terminating gateway.

a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy

!--- UDP redundancy is enabled.



!--- A trying message is sent and an !--- attempt is made to determine if T.38 fax-relay is supported.


*Mar 1 00:33:55.275: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1A735
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Fri, 01 Mar 2002 00:33:55 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow-Events: telephone-event
Remote-Party-ID: <sip:9000@172.16.13.87>;party=called;screen=no;privacy=off
Content-Length: 0


!--- The OK to the re-invite that specifies that you can !--- do T.38 fax-relay. The same UDP port is retained.

 
*Mar 1 00:33:55.275: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1A735
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Fri, 01 Mar 2002 00:33:55 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:9000@172.16.13.87>;party=called;screen=no;privacy=off
Contact: <sip:9000@172.16.13.87:5060>
Content-Type: application/sdp
Content-Length: 157

v=0
o=CiscoSystemsSIP-GW-UserAgent 1715 2725 IN IP4 172.16.13.87
s=SIP Call
c=IN IP4 172.16.13.87
t=0 0
m=image 18080 udptl t38
c=IN IP4 172.16.13.87


!--- The ack to the OK is received. At this point, fax transmission occurs. 

*Mar 1 00:33:55.719: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
ACK sip:9000@172.16.13.87:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1B21D0
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Tue, 28 Feb 2006 23:43:49 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0


!--- Once the fax transmission is completed, !--- the BYE is received. The BYE is similar to a !--- release message in Q.931.


*Mar 1 00:34:45.515: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
BYE sip:9000@172.16.13.87:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1E1E51
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Tue, 28 Feb 2006 23:44:38 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1141170279
CSeq: 103 BYE
Reason: Q.850;cause=16

!--- Cause code 16 is a normal disconnect cause.

Content-Length: 0

!--- There should be an OK to every message.

*Mar 1 00:34:45.535: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.184.83:5060;branch=z9hG4bK1E1E51
From: <sip:8000@172.16.184.83>;tag=85E9C7C8-A4C
To: <sip:9000@172.16.13.87>;tag=1EDC10-2436
Date: Fri, 01 Mar 2002 00:34:45 GMT
Call-ID: D110EA36-2BE211D6-801CEF21-DD62106B@172.16.13.87
Server: Cisco-SIPGateway/IOS-12.x
Timestamp: 1141170279
CSeq: 103 BYE
Reason: Q.850;cause=16
Content-Length: 0
 

More information about the attributes:

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)
z=* (time zone adjustments)
k=* (encryption key)
a=* (zero or more session attribute lines)

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)
* indicated optional item.

Basic Requests
INVITE: request from a UAC to initiate a session
ACK: confirms receipt of a final response to INVITE
BYE: sent by either side to end a session
CANCEL: sent to end a call not yet connected
UPDATE: Updates offer for not-yet-established sessions.
REGISTER: UA registers with Registrar Server
NOTIFY: notifies that an event has occurred
REFER: the mechanism to initiate a session transfer
INFO: a means of carrying ?data? in a message body


SIP responses:

1xx: Provisional ? request received, continuing to process the request 
2xx: Success - action was successfully received, understood, and accepted 
3xx: Redirection - further action needs to be taken in order to complete the request 
4xx: Client Error - the request contains bad syntax or cannot be 
fulfilled at this server 
5xx: Server Error - the server failed to fulfill an apparently valid request 
6xx: Global Failure - the request cannot be fulfilled at any server 

Informations connexes

• RFC SDP 2327
• SIP RFC 3261
• Assistance technique concernant la technologie vocale
• Assistance concernant les produits vocaux et de communications unifiées
• Dépannage des problèmes de téléphonie IP Cisco
• Support et documentation techniques - Cisco Systems