[asterisk-users] Headaches with Video over SIP
Peter Howard
peter.howard at ursys.com.au
Mon Nov 13 15:28:25 MST 2006
On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
> any logs/errors when you do a verbose 6 and a sip debug ?
>
I've got a log from a call under asterisk 1.4.0-beta3 attached. The
behaviour was the same; the call connected and audio worked, but no
video.
> On 11/13/06, Peter Howard <peter.howard at ursys.com.au> wrote:
> On Mon, 2006-11-13 at 00:57 +0100, Patrick wrote:
> > On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote:
> > > Greetings all,
> > >
> > > I'm playing with asterisk and two Polycom VSX300
> videoconferencing
> > > units. And I'm having zero luck getting video working
> over SIP.
> > >
> > > The two units register fine with asterisk, and with
> "allow=all" in
> > > sip.conf, the two units establish voice. But no
> video. And no obvious
> > > messages as to whats going wrong. The config for each is
> (they're
> > > numbered 201 and 202):
> > >
> > > [202]
> > > secret=
> > > type=friend
> > > context=from-sip-202
> > > host=dynamic
> > > nat=no
> > > canreinvite=yes
> > > dtmfmode=rfc2833
> > > disallow=all
> > > allow=all
> > >
> > >
> > > If you're wondering why I do the "disallow=all"
> immediately followed by
> > > "allow=all", it's because the allow line has spent a lot
> of time with
> > > restricted codecs to see if that makes a difference.
> > >
> > > I can provide the full sip.conf, extensions.conf, and
> debug output if
> > > anyone wants to see them.
> > >
> > > Any suggestions as to where things are falling down?
> >
> > Do you have "videosupport=yes" in your sip.conf?
>
> Yes I do. I've also confirmed that I have a version of
> asterisk which
> includes the patch for H263P (which is what the Polycoms want
> to talk).
>
> --
> Peter Howard
> URSYS
> 13 Burwood Rd,
> Burwood, NSW 2134
>
> Ph: 02 8745 2816 Fax: 02 8745 2828
>
> _______________________________________________
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--
Peter Howard
URSYS
13 Burwood Rd,
Burwood, NSW 2134
Ph: 02 8745 2816 Fax: 02 8745 2828
-------------- next part --------------
*CLI> sip list peers
Name/username Host Dyn Nat ACL Port Status
202 10.0.6.150 D 5060 Unmonitored
201 10.0.6.198 D 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
*CLI> sip show peer 201
* Name : 201
Secret : <Not set>
MD5Secret : <Not set>
Context : from-sip-201
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : -1
Call limit : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 57
Insecure : no
Nat : RFC3581
ACL : No
T38 pt UDPTL : No
T38 pt RTP : No
T38 pt TCP : No
CanReinvite : Yes
PromiscRedir : No
User=Phone : No
Video Support: Yes
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 10.0.6.198 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username:
SIP Options : (none)
Codecs : 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264)
Codec Order : (none)
Status : Unmonitored
Useragent : Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24
Reg. Contact : sip:10.0.6.198
*CLI> set verbose 9
Verbosity was 3 and is now 9
The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core verbose' instead.
*CLI> core verbose 9
Verbosity is at least 9
*CLI> core debug 9
Core debug was 0 and is now 9
*CLI> sip debug
SIP Debugging enabled
*CLI>
<--- SIP read from 10.0.6.198:5060 --->
INVITE sip:202 at 10.0.6.198 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5
Max-Forwards: 70
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1;tag=ec34af13
To: <sip:202 at 10.0.6.198>
Call-ID: 209bd28f at 10.0.6.198
CSeq: 1 INVITE
User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24
Contact: <sip:10.0.6.198>
Content-Type: application/sdp
Content-Length: 888
v=0
o=Ursys2 1468211940 0 IN IP4 10.0.6.198
s=-
c=IN IP4 10.0.6.198
b=AS:128
t=0 0
m=audio 49168 RTP/AVP 99 98 97 102 101 103 9 15 0 8 18
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:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729A/8000
m=video 49170 RTP/AVP 109 34 96 31
b=TIAS:128000
a=rtpmap:109 H264/90000
a=fmtp:109 profile-level-id=42800c max-mbps=10000
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2 F J T
a=rtpmap:31 H261/90000
a=fmtp:31 CIF=1 QCIF=1
m=data 49172 RTP/AVP 100
a=rtpmap:100 H224
<------------->
--- (11 headers 35 lines) ---
Sending to 10.0.6.198 : 5060 (no NAT)
Using INVITE request as basis request - 209bd28f at 10.0.6.198
Found user '201'
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 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP video format 109
Found RTP video format 34
Found RTP video format 96
Found RTP video format 31
[Nov 14 09:22:37] WARNING[11736]: chan_sip.c:4691 process_sdp: Unsupported SDP media type in offer: data 49172 RTP/AVP 100
Peer audio RTP is at port 10.0.6.198:49168
Found description format SIREN14 for ID 99
Got unsupported a:fmtp in SDP offer
Found description format SIREN14 for ID 98
Got unsupported a:fmtp in SDP offer
Found description format SIREN14 for ID 97
Got unsupported a:fmtp in SDP offer
Found description format G7221 for ID 102
Got unsupported a:fmtp in SDP offer
Found description format G7221 for ID 101
Got unsupported a:fmtp in SDP offer
Found description format G7221 for ID 103
Got unsupported a:fmtp in SDP offer
Found description format G722 for ID 9
Found description format G728 for ID 15
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729A for ID 18
Found description format H264 for ID 109
Got unsupported a:fmtp in SDP offer
Found description format H263 for ID 34
Found description format H263-1998 for ID 96
Got unsupported a:fmtp in SDP offer
Found description format H261 for ID 31
Got unsupported a:fmtp in SDP offer
Found description format H224 for ID 100
Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x3c050c (ulaw|alaw|g729|ilbc|h261|h263|h263p|h264)/video=0x3c0000 (h261|h263|h263p|h264), combined - 0x3c050c (ulaw|alaw|g729|ilbc|h261|h263|h263p|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.6.198:49168
Peer video RTP is at port 10.0.6.198:49170
Looking for 202 in from-sip-201 (domain 10.0.6.198)
list_route: hop: <sip:10.0.6.198>
<--- Transmitting (no NAT) to 10.0.6.198:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5;received=10.0.6.198
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1;tag=ec34af13
To: <sip:202 at 10.0.6.198>
Call-ID: 209bd28f at 10.0.6.198
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:202 at 10.0.6.11>
Content-Length: 0
<------------>
-- Executing [202 at from-sip-201:1] AGI("SIP/201-081e1340", "/etc/asterisk/setcid.agi|201") in new stack
-- Launched AGI Script /etc/asterisk/setcid.agi
== /etc/asterisk/setcid.agi|201: Failed to execute '/etc/asterisk/setcid.agi': No such file or directory
-- AGI Script /etc/asterisk/setcid.agi completed, returning 0
-- Executing [202 at from-sip-201:2] Goto("SIP/201-081e1340", "from-sip-cid-201|202|1") in new stack
-- Goto (from-sip-cid-201,202,1)
-- Executing [202 at from-sip-cid-201:1] Macro("SIP/201-081e1340", "general-dial|SIP/202") in new stack
-- Executing [s at macro-general-dial:1] Dial("SIP/201-081e1340", "SIP/202|20") in new stack
Video is at 10.0.6.11 port 12394
Audio is at 10.0.6.11 port 11736
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x40000 (h261) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.6.150:5060:
INVITE sip:10.0.6.150 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK2abc3e4a;rport
From: "201" <sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>
Contact: <sip:201 at 10.0.6.11>
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 13 Nov 2006 22:22:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 715
v=0
o=root 11716 11716 IN IP4 10.0.6.198
s=session
c=IN IP4 10.0.6.198
b=CT:384
t=0 0
m=audio 49168 RTP/AVP 0 4 3 8 112 5 10 7 18 110 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
m=video 49170 RTP/AVP 31 34 103 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
-- Called 202
<--- SIP read from 10.0.6.150:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK2abc3e4a;rport
From: "201"<sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>;tag=8548d49a
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 102 INVITE
Contact: <sip:10.0.6.150>
User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/202-081e7ca0 is ringing
<--- Transmitting (no NAT) to 10.0.6.198:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5;received=10.0.6.198
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1;tag=ec34af13
To: <sip:202 at 10.0.6.198>;tag=as2593d0f5
Call-ID: 209bd28f at 10.0.6.198
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:202 at 10.0.6.11>
Content-Length: 0
<------------>
<--- SIP read from 10.0.6.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK2abc3e4a;rport
From: "201"<sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>;tag=8548d49a
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 102 INVITE
Contact: <sip:10.0.6.150>
User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24
Content-Length: 425
Content-Type: application/sdp
v=0
o=URSYS 701409664 0 IN IP4 10.0.6.150
s=-
c=IN IP4 10.0.6.150
b=AS:128
t=0 0
m=audio 49160 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
m=video 49162 RTP/AVP 31 34 96 109
b=TIAS:128000
a=rtpmap:31 H261/90000
a=fmtp:31 CIF=1 QCIF=1
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2
a=rtpmap:109 H264/90000
a=fmtp:109 profile-level-id=42800c max-mbps=10000
<------------->
--- (10 headers 18 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP video format 31
Found RTP video format 34
Found RTP video format 96
Found RTP video format 109
Peer audio RTP is at port 10.0.6.150:49160
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format H261 for ID 31
Got unsupported a:fmtp in SDP offer
Found description format H263 for ID 34
Found description format H263-1998 for ID 96
Got unsupported a:fmtp in SDP offer
Found description format H264 for ID 109
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x3c000c (ulaw|alaw|h261|h263|h263p|h264)/video=0x3c0000 (h261|h263|h263p|h264), combined - 0x3c000c (ulaw|alaw|h261|h263|h263p|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.0.6.150:49160
Peer video RTP is at port 10.0.6.150:49162
list_route: hop: <sip:10.0.6.150>
set_destination: Parsing <sip:10.0.6.150> for address/port to send to
set_destination: set destination to 10.0.6.150, port 5060
Transmitting (no NAT) to 10.0.6.150:5060:
ACK sip:10.0.6.150 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK66b06791;rport
From: "201" <sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>;tag=8548d49a
Contact: <sip:201 at 10.0.6.11>
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/202-081e7ca0 answered SIP/201-081e1340
Video is at 10.0.6.11 port 14306
Audio is at 10.0.6.11 port 11746
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x40000 (h261) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.0.6.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5;received=10.0.6.198
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1;tag=ec34af13
To: <sip:202 at 10.0.6.198>;tag=as2593d0f5
Call-ID: 209bd28f at 10.0.6.198
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:202 at 10.0.6.11>
Content-Type: application/sdp
Content-Length: 500
v=0
o=root 11716 11716 IN IP4 10.0.6.150
s=session
c=IN IP4 10.0.6.150
b=CT:384
t=0 0
m=audio 49160 RTP/AVP 0 8 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
m=video 49162 RTP/AVP 31 34 103 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
<------------>
-- Native bridging SIP/201-081e1340 and SIP/202-081e7ca0
Retransmitting #1 (no NAT) to 10.0.6.198:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5;received=10.0.6.198
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1;tag=ec34af13
To: <sip:202 at 10.0.6.198>;tag=as2593d0f5
Call-ID: 209bd28f at 10.0.6.198
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:202 at 10.0.6.11>
Content-Type: application/sdp
Content-Length: 500
v=0
o=root 11716 11716 IN IP4 10.0.6.150
s=session
c=IN IP4 10.0.6.150
b=CT:384
t=0 0
m=audio 49160 RTP/AVP 0 8 18 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
m=video 49162 RTP/AVP 31 34 103 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
<--- SIP read from 10.0.6.198:5060 --->
ACK sip:10.0.6.198 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bKa052b6d5
Max-Forwards: 70
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1;tag=ec34af13
To: <sip:202 at 10.0.6.198>;tag=as2593d0f5
Call-ID: 209bd28f at 10.0.6.198
CSeq: 1 ACK
Contact: <sip:10.0.6.198>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from 10.0.6.198:5060 --->
REGISTER sip:10.0.6.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.198
Max-Forwards: 70
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1
To: <sip:201 at 10.0.6.198>
Call-ID: 448690996 at 10.0.6.198
CSeq: 482 REGISTER
User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24
Contact: <sip:10.0.6.198>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.0.6.198 : 5060 (no NAT)
<--- Transmitting (no NAT) to 10.0.6.198:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.6.198;received=10.0.6.198
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1
To: <sip:201 at 10.0.6.198>
Call-ID: 448690996 at 10.0.6.198
CSeq: 482 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:201 at 10.0.6.11>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.0.6.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.6.198;received=10.0.6.198
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1
To: <sip:201 at 10.0.6.198>;tag=as21d351c8
Call-ID: 448690996 at 10.0.6.198
CSeq: 482 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: <sip:10.0.6.198>;expires=120
Date: Mon, 13 Nov 2006 22:22:45 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '448690996 at 10.0.6.198' in 32000 ms (Method: REGISTER)
<--- SIP read from 10.0.6.198:5060 --->
BYE sip:202 at 10.0.6.198 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bK595bd6ab
Max-Forwards: 70
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1;tag=ec34af13
To: <sip:202 at 10.0.6.198>;tag=as2593d0f5
Call-ID: 209bd28f at 10.0.6.198
CSeq: 2 BYE
User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24
Contact: <sip:10.0.6.198>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 10.0.6.198 : 5060 (no NAT)
<--- Transmitting (no NAT) to 10.0.6.198:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.6.198;branch=z9hG4bK595bd6ab;received=10.0.6.198
From: <sip:201 at 10.0.6.198>;epid=82042503F512B1;tag=ec34af13
To: <sip:202 at 10.0.6.198>;tag=as2593d0f5
Call-ID: 209bd28f at 10.0.6.198
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:202 at 10.0.6.11>
Content-Length: 0
<------------>
set_destination: Parsing <sip:10.0.6.150> for address/port to send to
set_destination: set destination to 10.0.6.150, port 5060
Video is at 10.0.6.11 port 12394
Audio is at 10.0.6.11 port 11736
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x40000 (h261) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding codec 0x200000 (h264) to SDP
Reliably Transmitting (no NAT) to 10.0.6.150:5060:
INVITE sip:10.0.6.150 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK59b84b97;rport
From: "201" <sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>;tag=8548d49a
Contact: <sip:201 at 10.0.6.11>
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 350
v=0
o=root 11716 11717 IN IP4 10.0.6.11
s=session
c=IN IP4 10.0.6.11
b=CT:384
t=0 0
m=audio 11736 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=sendrecv
m=video 12394 RTP/AVP 31 34 103 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
Scheduling destruction of SIP dialog '7019698d5791958b1c1812be36c3dfcc at 10.0.6.11' in 32000 ms (Method: INVITE)
== Spawn extension (macro-general-dial, s, 1) exited non-zero on 'SIP/201-081e1340'
-- Executing [h at macro-general-dial:1] Hangup("SIP/201-081e1340", "") in new stack
== Spawn extension (macro-general-dial, h, 1) exited non-zero on 'SIP/201-081e1340'
<--- SIP read from 10.0.6.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK59b84b97;rport
From: "201"<sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>;tag=8548d49a
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 103 INVITE
Contact: <sip:10.0.6.150>
User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24
Content-Length: 422
Content-Type: application/sdp
v=0
o=URSYS 1279946760 0 IN IP4 10.0.6.150
s=-
c=IN IP4 10.0.6.150
b=AS:128
t=0 0
m=audio 49160 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
m=video 0 RTP/AVP 31 34 96 109
b=TIAS:128000
a=rtpmap:31 H261/90000
a=fmtp:31 CIF=1 QCIF=1
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2
a=rtpmap:109 H264/90000
a=fmtp:109 profile-level-id=42800c max-mbps=10000
<------------->
--- (10 headers 18 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP video format 31
Found RTP video format 34
Found RTP video format 96
Found RTP video format 109
Peer audio RTP is at port 10.0.6.150:49160
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format H261 for ID 31
Got unsupported a:fmtp in SDP offer
Found description format H263 for ID 34
Found description format H263-1998 for ID 96
Got unsupported a:fmtp in SDP offer
Found description format H264 for ID 109
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x3c000c (ulaw|alaw|h261|h263|h263p|h264)/video=0x3c0000 (h261|h263|h263p|h264), combined - 0x3c000c (ulaw|alaw|h261|h263|h263p|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.0.6.150:49160
set_destination: Parsing <sip:10.0.6.150> for address/port to send to
set_destination: set destination to 10.0.6.150, port 5060
Transmitting (no NAT) to 10.0.6.150:5060:
ACK sip:10.0.6.150 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK06f3c449;rport
From: "201" <sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>;tag=8548d49a
Contact: <sip:201 at 10.0.6.11>
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
set_destination: Parsing <sip:10.0.6.150> for address/port to send to
set_destination: set destination to 10.0.6.150, port 5060
Reliably Transmitting (no NAT) to 10.0.6.150:5060:
BYE sip:10.0.6.150 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK2a2fe0b0;rport
From: "201" <sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>;tag=8548d49a
Contact: <sip:201 at 10.0.6.11>
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog '7019698d5791958b1c1812be36c3dfcc at 10.0.6.11' in 32000 ms (Method: INVITE)
Really destroying SIP dialog '209bd28f at 10.0.6.198' Method: BYE
<--- SIP read from 10.0.6.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK59b84b97;rport
From: "201"<sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>;tag=8548d49a
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 103 INVITE
Contact: <sip:10.0.6.150>
User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24
Content-Length: 422
Content-Type: application/sdp
v=0
o=URSYS 1279946760 0 IN IP4 10.0.6.150
s=-
c=IN IP4 10.0.6.150
b=AS:128
t=0 0
m=audio 49160 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
m=video 0 RTP/AVP 31 34 96 109
b=TIAS:128000
a=rtpmap:31 H261/90000
a=fmtp:31 CIF=1 QCIF=1
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=fmtp:96 SQCIF=1 QCIF=1 CIF=1 CIF4=2
a=rtpmap:109 H264/90000
a=fmtp:109 profile-level-id=42800c max-mbps=10000
<------------->
--- (10 headers 18 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP video format 31
Found RTP video format 34
Found RTP video format 96
Found RTP video format 109
Peer audio RTP is at port 10.0.6.150:49160
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format H261 for ID 31
Got unsupported a:fmtp in SDP offer
Found description format H263 for ID 34
Found description format H263-1998 for ID 96
Got unsupported a:fmtp in SDP offer
Found description format H264 for ID 109
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x3c000c (ulaw|alaw|h261|h263|h263p|h264)/video=0x3c0000 (h261|h263|h263p|h264), combined - 0x3c000c (ulaw|alaw|h261|h263|h263p|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.0.6.150:49160
set_destination: Parsing <sip:10.0.6.150> for address/port to send to
set_destination: set destination to 10.0.6.150, port 5060
Transmitting (no NAT) to 10.0.6.150:5060:
ACK sip:10.0.6.150 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK566237bd;rport
From: "201" <sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>;tag=8548d49a
Contact: <sip:201 at 10.0.6.11>
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<--- SIP read from 10.0.6.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.6.11:5060;branch=z9hG4bK2a2fe0b0;rport
From: "201"<sip:201 at 10.0.6.11>;tag=as79c5ae7b
To: <sip:10.0.6.150>;tag=8548d49a
Call-ID: 7019698d5791958b1c1812be36c3dfcc at 10.0.6.11
CSeq: 104 BYE
Contact: <sip:10.0.6.150>
User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '7019698d5791958b1c1812be36c3dfcc at 10.0.6.11' Method: INVITE
Really destroying SIP dialog '448690996 at 10.0.6.198' Method: REGISTER
<--- SIP read from 10.0.6.150:5060 --->
REGISTER sip:10.0.6.11 SIP/2.0
Via: SIP/2.0/UDP 10.0.6.150
Max-Forwards: 70
From: <sip:202 at 10.0.6.150>;epid=82042503F811B1
To: <sip:202 at 10.0.6.150>
Call-ID: 663860435 at 10.0.6.150
CSeq: 493 REGISTER
User-Agent: Polycom VSX 3000 Release 7.0 - 10Jul2004 01:24
Contact: <sip:10.0.6.150>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.0.6.150 : 5060 (no NAT)
<--- Transmitting (no NAT) to 10.0.6.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.6.150;received=10.0.6.150
From: <sip:202 at 10.0.6.150>;epid=82042503F811B1
To: <sip:202 at 10.0.6.150>
Call-ID: 663860435 at 10.0.6.150
CSeq: 493 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:202 at 10.0.6.11>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.0.6.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.6.150;received=10.0.6.150
From: <sip:202 at 10.0.6.150>;epid=82042503F811B1
To: <sip:202 at 10.0.6.150>;tag=as6806220b
Call-ID: 663860435 at 10.0.6.150
CSeq: 493 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: <sip:10.0.6.150>;expires=120
Date: Mon, 13 Nov 2006 22:23:39 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '663860435 at 10.0.6.150' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '663860435 at 10.0.6.150' Method: REGISTER
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