[asterisk-bugs] [Asterisk 0010782]: Video doesn't work for outgoing call?
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri Sep 21 01:14:03 CDT 2007
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=10782
======================================================================
Reported By: cwhuang
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 10782
Category: Channels/chan_sip/Video
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.4.11
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 83398
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 09-21-2007 01:03 CDT
Last Modified: 09-21-2007 01:14 CDT
======================================================================
Summary: Video doesn't work for outgoing call?
Description:
I've tried to put a call file to /var/spool/asterisk/outgoing/
to make an outgoing video call, but not succeeded.
I could hear the audio, but no video.
I have tried both asterisk 1.4.11 and svn branches 1.4.
Both don't work.
The client is Leadtek BVP8882 video phone, with H263 codec.
Here are some debug messages.
It shows the client and asterisk negotiated the video capabilities
without problem. However, the 'show channel ...' command
shows there is only audio channel, no video channel.
I don't understand why.
The file I tried to play is
$ ls -al /var/lib/asterisk/VOD/jolin-512k*
-rw-r--r-- 1 root root 7975341 2007-03-09 13:16
/var/lib/asterisk/VOD/jolin-512k.gsm
-rw-r--r-- 1 root root 236916736 2007-03-09 13:16
/var/lib/asterisk/VOD/jolin-512k.h263
both .gsm and .h263 are available.
I'm sure the media files have no problem,
since I can see the video by making a call from
video phone to asterisk.
I have also tried to make outgoing call by the manager API.
There is no video, either.
cwhuang*CLI> sip debug peer 405
SIP Debugging Enabled for IP: 10.10.130.51:5060
The 'sip debug' command is deprecated and will be removed in a future
release. Please use 'sip set debug' instead.
Video is at 10.10.130.55 port 17032
Audio is at 10.10.130.55 port 17300
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.10.130.51:5060:
INVITE sip:405 at 10.10.130.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.130.55:5060;branch=z9hG4bK278c9e5b;rport
From: "555" <sip:555 at 10.10.130.55>;tag=as3b33ee0c
To: <sip:405 at 10.10.130.51:5060>
Contact: <sip:555 at 10.10.130.55>
Call-ID: 54a6dda52c01533878c7a9ce0584806f at 10.10.130.55
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 21 Sep 2007 06:10:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 339
v=0
o=root 17835 17835 IN IP4 10.10.130.55
s=session
c=IN IP4 10.10.130.55
b=CT:384
t=0 0
m=audio 17300 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 17032 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
Retransmitting http://bugs.digium.com/view.php?id=1 (no NAT) to
10.10.130.51:5060:
INVITE sip:405 at 10.10.130.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.130.55:5060;branch=z9hG4bK278c9e5b;rport
From: "555" <sip:555 at 10.10.130.55>;tag=as3b33ee0c
To: <sip:405 at 10.10.130.51:5060>
Contact: <sip:555 at 10.10.130.55>
Call-ID: 54a6dda52c01533878c7a9ce0584806f at 10.10.130.55
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 21 Sep 2007 06:10:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 339
v=0
o=root 17835 17835 IN IP4 10.10.130.55
s=session
c=IN IP4 10.10.130.55
b=CT:384
t=0 0
m=audio 17300 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 17032 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
cwhuang*CLI>
<--- SIP read from 10.10.130.51:5060 --->
SIP/2.0 100 Trying
Call-ID: 54a6dda52c01533878c7a9ce0584806f at 10.10.130.55
From: 555<sip:555 at 10.10.130.55>;tag=as3b33ee0c
To: <sip:405 at 10.10.130.51:5060>;tag=10007900-5cc3498
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.10.130.55:5060;branch=z9hG4bK278c9e5b
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
cwhuang*CLI>
<--- SIP read from 10.10.130.51:5060 --->
SIP/2.0 100 Trying
Call-ID: 54a6dda52c01533878c7a9ce0584806f at 10.10.130.55
From: 555<sip:555 at 10.10.130.55>;tag=as3b33ee0c
To: <sip:405 at 10.10.130.51:5060>;tag=10007900-5cc3498
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.10.130.55:5060;branch=z9hG4bK278c9e5b
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
cwhuang*CLI>
<--- SIP read from 10.10.130.51:5060 --->
SIP/2.0 180 Ringing
Call-ID: 54a6dda52c01533878c7a9ce0584806f at 10.10.130.55
From: 555<sip:555 at 10.10.130.55>;tag=as3b33ee0c
To: <sip:405 at 10.10.130.51:5060>;tag=10007900-5cc3498
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.10.130.55:5060;branch=z9hG4bK278c9e5b
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
cwhuang*CLI>
<--- SIP read from 10.10.130.51:5060 --->
SIP/2.0 200 OK
Call-ID: 54a6dda52c01533878c7a9ce0584806f at 10.10.130.55
From: 555<sip:555 at 10.10.130.55>;tag=as3b33ee0c
To: <sip:405 at 10.10.130.51:5060>;tag=10007900-5cc3498
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.10.130.55:5060;branch=z9hG4bK278c9e5b
Contact: sip:405 at 10.10.130.51:5060
Allow: REFER,INFO,NOTIFY
User-Agent: GVSC LR8882 9.9.99_57
Content-Type: application/sdp
Content-Length: 233
v=0
o=405 4352 4352 IN IP4 10.10.130.51
s=-
c=IN IP4 10.10.130.51
t=0 0
m=audio 8050 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 8060 RTP/AVP 34
b=AS:192
a=rtpmap:34 H263/90000
<------------->
--- (11 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found RTP video format 34
Peer audio RTP is at port 10.10.130.51:8050
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Found description format H263 for ID 34
Capabilities: us - 0x280004 (ulaw|h263|h264), peer - audio=0x80004
(ulaw|h263)/video=0x80000 (h263), combined - 0x80004 (ulaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.10.130.51:8050
Peer video RTP is at port 10.10.130.51:8060
list_route: hop: <sip:405 at 10.10.130.51:5060>
set_destination: Parsing <sip:405 at 10.10.130.51:5060> for address/port to
send to
set_destination: set destination to 10.10.130.51, port 5060
Transmitting (no NAT) to 10.10.130.51:5060:
ACK sip:405 at 10.10.130.51:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.130.55:5060;branch=z9hG4bK72bf4ac8;rport
From: "555" <sip:555 at 10.10.130.55>;tag=as3b33ee0c
To: <sip:405 at 10.10.130.51:5060>;tag=10007900-5cc3498
Contact: <sip:555 at 10.10.130.55>
Call-ID: 54a6dda52c01533878c7a9ce0584806f at 10.10.130.55
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
cwhuang*CLI> show channel SIP/405-094e6990
-- General --
Name: SIP/405-094e6990
Type: SIP
UniqueID: 1190355046.1
Caller ID: 555
Caller ID Name: (N/A)
DNID Digits: (N/A)
State: Up (6)
Rings: 0
NativeFormats: 0x4 (ulaw)
WriteFormat: 0x2 (gsm)
ReadFormat: 0x4 (ulaw)
WriteTranscode: Yes
ReadTranscode: No
1st File Descriptor: 39
Frames in: 382
Frames out: 439
Time to Hangup: 0
Elapsed Time: 0h0m15s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: macro-play
Extension: s
Priority: 3
Call Group: 0
Pickup Group: 0
Application: BackGround
Data: /var/lib/asterisk/VOD/jolin-512k
Blocking in: ast_waitfor_nandfds
Variables:
MACRO_DEPTH=2
ARG1=/var/lib/asterisk/VOD/jolin-512k
MACRO_PRIORITY=5
MACRO_CONTEXT=macro-playvod
MACRO_EXTEN=s
FROM_IVR=yes
MOIVE=jolin-512k
RECORDED=broadcast/msg-24
SIPCALLID=54a6dda52c01533878c7a9ce0584806f at 10.10.130.55
CDR Variables:
level 1: clid=555
level 1: src=555
level 1: dst=t
level 1: dcontext=vod
level 1: channel=SIP/405-094e6990
level 1: lastapp=BackGround
level 1: lastdata=/var/lib/asterisk/VOD/jolin-512k
level 1: start=2007-09-21 14:10:46
level 1: answer=2007-09-21 14:10:52
level 1: end=2007-09-21 14:10:52
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1190355046.1
======================================================================
----------------------------------------------------------------------
cwhuang - 09-21-07 01:14
----------------------------------------------------------------------
I just tried the svn trunk (rev 83396).
It works. Amazing...
So there must be a change that fix this issue.
However, I'm not going to use trunk, since it seems still not stable for a
production usage. For example, every time I reload the pbx, it core
dumped.
Could you investigate the difference and backport the change to branch
1.4?
The show channel command shows there is a video channel.
*CLI> core show channel SIP/405-098bf728
-- General --
Name: SIP/405-098bf728
Type: SIP
UniqueID: 1190355937.0
Caller ID: 555
Caller ID Name: (N/A)
DNID Digits: (N/A)
Language: en
State: Up (6)
Rings: 0
NativeFormats: 0x80004 (ulaw|h263)
WriteFormat: 0x2 (gsm)
ReadFormat: 0x40 (slin)
WriteTranscode: Yes
ReadTranscode: Yes
1st File Descriptor: 37
Frames in: 735
Frames out: 1511
Time to Hangup: 0
Elapsed Time: 0h0m19s
Direct Bridge: <none>
Indirect Bridge: <none>
-- PBX --
Context: macro-play
Extension: s
Priority: 3
Call Group: 0
Pickup Group: 0
Application: BackGround
Data: /var/lib/asterisk/VOD/jolin-512k
Blocking in: ast_waitfor_nandfds_simple
Variables:
MACRO_DEPTH=2
ARG1=/var/lib/asterisk/VOD/jolin-512k
MACRO_PRIORITY=5
MACRO_CONTEXT=macro-playvod
MACRO_EXTEN=s
FROM_IVR=yes
MOIVE=jolin-512k
RECORDED=broadcast/msg-24
SIPCALLID=5a8532026f00d20a4937e6896d4a39f7 at 10.10.130.55
CDR Variables:
level 1: clid=555
level 1: src=555
level 1: dst=t
level 1: dcontext=vod
level 1: channel=SIP/405-098bf728
level 1: lastapp=BackGround
level 1: lastdata=/var/lib/asterisk/VOD/jolin-512k
level 1: start=2007-09-21 14:25:37
level 1: answer=2007-09-21 14:25:42
level 1: end=1970-01-01 08:00:00
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1190355937.0
Issue History
Date Modified Username Field Change
======================================================================
09-21-07 01:14 cwhuang Note Added: 0070887
======================================================================
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