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09.03.2008 at 07:16AM PDT, ID: 23699268
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8.2

SIP Call between two Polycom PVX endpoints using Asterisk

Asked by WideAreaMedia in Asterisk Open Source Telephony, Voice Over IP, IP PBX Systems

Tags: , , ,

Hi Everyone,

I'm trying to place a video call between two Polycom PVX clients registered using SIP. Both are registered properly. When I attempt to call, the far endpoint does not ring. I get the following messages in my CLI console:

== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
[Sep 2 16:04:30] WARNING[25440]: chan_sip.c:6502 process_sdp: Unsupported SDP media type in offer: application 3234 RTP/AVP 100
[Sep 2 16:04:30] NOTICE[25440]: chan_sip.c:16416 handle_request_invite: Call from '1001' to extension '1002' rejected because extension not found.

I've no idea what the problem is, but I'm guessing perhaps a codec needs to be enabled. I'm very new to this, so any help would be appreciated. Thanks!

Best Regards,
Martin Schultz

The SIP debug messages are:

sip set debug on
SIP Debugging enabled
*CLI>
<--- SIP read from TCP://68.109.238.117:53533 --->
INVITE sip:1002 SIP/2.0
Via: SIP/2.0/TCP 68.109.238.117:5060;branch=z9hG4bK2737850136-18
Max-Forwards: 70
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
Supported: timer,replaces,100rel
From: 1001<sip:1001@68.109.238.117>;epid=87947d54-a9c;tag=plcm_2737850136-19
To: <sip:1002>
Call-ID: 2737850136-17
CSeq: 1 INVITE
Session-Expires: 600
Contact: <sip:68.109.238.117:5060;transport=tcp>
User-Agent: Polycom VV 8.0.4.4035
Content-Type: application/sdp
Content-Length: 1019

v=0
o=1001 12065 0 IN IP4 68.109.238.117
s=-
c=IN IP4 68.109.238.117
b=AS:384
t=0 0
m=audio 3230 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8
a=rtpmap:99 SIREN14/16000
a=fmtp:99 bitrate=48000
a=rtpmap:98 SIREN14/16000
a=fmtp:98 bitrate=32000
a=rtpmap:97 SIREN14/16000
a=fmtp:97 bitrate=24000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:101 G7221/16000
a=fmtp:101 bitrate=24000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=16000
a=rtpmap:9 G722/8000
a=rtpmap:15 G728/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
m=video 3232 RTP/AVP 109 34 96 31
b=TIAS:384000
a=rtpmap:109 H264/90000
a=fmtp:109 profile-level-id=42800d; max-mbps=40000; max-fs=1792; max-br=1025
a=rtpmap:34 H263/90000
a=fmtp:34 CIF4=1;CIF=1;QCIF=1;SQCIF=1;F
a=rtpmap:96 H263-1998/90000
a=fmtp:96 CIF4=1;CIF=1;QCIF=1;SQCIF=1;F;J;T
a=rtpmap:31 H261/90000
a=fmtp:31 CIF=1;QCIF=1
a=sendrecv
m=application 3234 RTP/AVP 100
a=rtpmap:100 H224/0
a=sendrecv

<------------->
--- (14 headers 40 lines) ---
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
Sending to 68.109.238.117 : 5060 (no NAT)
Using INVITE request as basis request - 2737850136-17
Found user '1001' for '1001'

<--- Reliably Transmitting (no NAT) to 68.109.238.117:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 68.109.238.117:5060;branch=z9hG4bK2737850136-18;received=68.109.238.117
From: 1001<sip:1001@68.109.238.117>;epid=87947d54-a9c;tag=plcm_2737850136-19
To: <sip:1002>;tag=as7e35c0e8
Call-ID: 2737850136-17
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4fc4e8a8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2737850136-17' in 32000 ms (Method: INVITE)

<--- SIP read from TCP://68.109.238.117:53533 --->
ACK sip:1002 SIP/2.0
Via: SIP/2.0/TCP 68.109.238.117:5060;branch=z9hG4bK2737850136-18
Max-Forwards: 70
From: 1001<sip:1001@68.109.238.117>;epid=87947d54-a9c;tag=plcm_2737850136-19
To: <sip:1002>;tag=as7e35c0e8
Call-ID: 2737850136-17
CSeq: 1 ACK
Contact: <sip:68.109.238.117:5060;transport=tcp>
User-Agent: Polycom VV 8.0.4.4035
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from TCP://68.109.238.117:53533 --->
INVITE sip:1002 SIP/2.0
Via: SIP/2.0/TCP 68.109.238.117:5060;branch=z9hG4bK2738069136-20
Max-Forwards: 70
From: 1001<sip:1001@68.109.238.117>;epid=87947d54-a9c;tag=plcm_2737850136-19
To: <sip:1002>
Call-ID: 2737850136-17
CSeq: 2 INVITE
Session-Expires: 600
Supported: timer
Contact: <sip:68.109.238.117:5060;transport=tcp>
Content-Type: application/sdp
Authorization: Digest username="1001",realm="asterisk",nonce="4fc4e8a8",uri="sip:1002",response="d03b1534f465d3f3d6ed0133db27b955",algorithm=MD5
User-Agent: Polycom VV 8.0.4.4035
Content-Length: 1019

v=0
o=1001 12065 0 IN IP4 68.109.238.117
s=-
c=IN IP4 68.109.238.117
b=AS:384
t=0 0
m=audio 3230 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8
a=rtpmap:99 SIREN14/16000
a=fmtp:99 bitrate=48000
a=rtpmap:98 SIREN14/16000
a=fmtp:98 bitrate=32000
a=rtpmap:97 SIREN14/16000
a=fmtp:97 bitrate=24000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:101 G7221/16000
a=fmtp:101 bitrate=24000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=16000
a=rtpmap:9 G722/8000
a=rtpmap:15 G728/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
m=video 3232 RTP/AVP 109 34 96 31
b=TIAS:384000
a=rtpmap:109 H264/90000
a=fmtp:109 profile-level-id=42800d; max-mbps=40000; max-fs=1792; max-br=1025
a=rtpmap:34 H263/90000
a=fmtp:34 CIF4=1;CIF=1;QCIF=1;SQCIF=1;F
a=rtpmap:96 H263-1998/90000
a=fmtp:96 CIF4=1;CIF=1;QCIF=1;SQCIF=1;F;J;T
a=rtpmap:31 H261/90000
a=fmtp:31 CIF=1;QCIF=1
a=sendrecv
m=application 3234 RTP/AVP 100
a=rtpmap:100 H224/0
a=sendrecv

<------------->
--- (14 headers 40 lines) ---
Sending to 68.109.238.117 : 5060 (no NAT)
Using INVITE request as basis request - 2737850136-17
Found user '1001' for '1001'
Found RTP audio format 99
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 102
Found RTP audio format 101
Found RTP audio format 103
Found RTP audio format 9
Found RTP audio format 15
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP video format 109
Found RTP video format 34
Found RTP video format 96
Found RTP video format 31
[Sep 2 16:06:56] WARNING[30473]: chan_sip.c:6502 process_sdp: Unsupported SDP media type in offer: application 3234 RTP/AVP 100
Peer audio RTP is at port 68.109.238.117:3230
Found unknown media description format SIREN14 for ID 99
Found unknown media description format SIREN14 for ID 98
Found unknown media description format SIREN14 for ID 97
Found unknown media description format G7221 for ID 102
Found unknown media description format G7221 for ID 101
Found unknown media description format G7221 for ID 103
Found audio description format G722 for ID 9
Found unknown media description format G728 for ID 15
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found video description format H264 for ID 109
Found video description format H263 for ID 34
Found video description format H263-1998 for ID 96
Found video description format H261 for ID 31
Found unknown media description format H224 for ID 100
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x3c0000 (h261|h263|h263p|h264)/text=0x0 (nothing), combined - 0x8000c (ulaw|alaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 68.109.238.117:3230
Peer video RTP is at port 68.109.238.117:3232
Looking for s in phones (domain 1002)

<--- Reliably Transmitting (no NAT) to 68.109.238.117:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 68.109.238.117:5060;branch=z9hG4bK2738069136-20;received=68.109.238.117
From: 1001<sip:1001@68.109.238.117>;epid=87947d54-a9c;tag=plcm_2737850136-19
To: <sip:1002>;tag=as7e35c0e8
Call-ID: 2737850136-17
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
[Sep 2 16:06:56] NOTICE[30473]: chan_sip.c:16416 handle_request_invite: Call from '1001' to extension '1002' rejected because extension not found.
Scheduling destruction of SIP dialog '2737850136-17' in 32000 ms (Method: INVITE)

<--- SIP read from TCP://68.109.238.117:53533 --->
ACK sip:1002 SIP/2.0
Via: SIP/2.0/TCP 68.109.238.117:5060;branch=z9hG4bK2738069136-20
Max-Forwards: 70
Authorization: Digest username="1001",realm="asterisk",nonce="4fc4e8a8",uri="sip:1002",response="d03b1534f465d3f3d6ed0133db27b955",algorithm=MD5
From: 1001<sip:1001@68.109.238.117>;epid=87947d54-a9c;tag=plcm_2737850136-19
To: <sip:1002>;tag=as7e35c0e8
Call-ID: 2737850136-17
CSeq: 2 ACK
Contact: <sip:68.109.238.117:5060;transport=tcp>
User-Agent: Polycom VV 8.0.4.4035
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---Start Free Trial
[+][-]09.04.2008 at 01:32AM PDT, ID: 22385126

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About this solution

Zones: Asterisk Open Source Telephony, Voice Over IP, IP PBX Systems
Tags: Digium, Asterisk, 1.6-Beta9, RHEL Linux Host
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Solution Provided By: feptias
Participating Experts: 2
Solution Grade: A
 
 
[+][-]09.11.2008 at 04:32PM PDT, ID: 22456029

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