Summary: | ASTERISK-10355: Video doesn't work for outgoing call? | ||
Reporter: | Chih-Wei Huang (cwhuang) | Labels: | |
Date Opened: | 2007-09-21 01:03:04 | Date Closed: | 2007-09-21 08:17:09 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/Video |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
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@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@10.10.130.55>;tag=as3b33ee0c To: <sip:405@10.10.130.51:5060> Contact: <sip:555@10.10.130.55> Call-ID: 54a6dda52c01533878c7a9ce0584806f@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 #1 (no NAT) to 10.10.130.51:5060: INVITE sip:405@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@10.10.130.55>;tag=as3b33ee0c To: <sip:405@10.10.130.51:5060> Contact: <sip:555@10.10.130.55> Call-ID: 54a6dda52c01533878c7a9ce0584806f@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@10.10.130.55 From: 555<sip:555@10.10.130.55>;tag=as3b33ee0c To: <sip:405@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@10.10.130.55 From: 555<sip:555@10.10.130.55>;tag=as3b33ee0c To: <sip:405@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@10.10.130.55 From: 555<sip:555@10.10.130.55>;tag=as3b33ee0c To: <sip:405@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@10.10.130.55 From: 555<sip:555@10.10.130.55>;tag=as3b33ee0c To: <sip:405@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@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@10.10.130.51:5060> set_destination: Parsing <sip:405@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@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@10.10.130.55>;tag=as3b33ee0c To: <sip:405@10.10.130.51:5060>;tag=10007900-5cc3498 Contact: <sip:555@10.10.130.55> Call-ID: 54a6dda52c01533878c7a9ce0584806f@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@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 | ||
Comments: | By: Chih-Wei Huang (cwhuang) 2007-09-21 01:14:03 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@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 By: Digium Subversion (svnbot) 2007-09-21 08:15:41 Repository: asterisk Revision: 83400 ------------------------------------------------------------------------ r83400 | file | 2007-09-21 08:15:40 -0500 (Fri, 21 Sep 2007) | 4 lines Fix video under certain circumstances. It would have been possible for the formats on the channel to not contain the video format. (closes issue ASTERISK-10355) Reported by: cwhuang ------------------------------------------------------------------------ By: Digium Subversion (svnbot) 2007-09-21 08:17:09 Repository: asterisk Revision: 83401 ------------------------------------------------------------------------ r83401 | file | 2007-09-21 08:17:09 -0500 (Fri, 21 Sep 2007) | 11 lines Blocked revisions 83400 via svnmerge ........ r83400 | file | 2007-09-21 10:34:32 -0300 (Fri, 21 Sep 2007) | 4 lines Fix video under certain circumstances. It would have been possible for the formats on the channel to not contain the video format. (closes issue ASTERISK-10355) Reported by: cwhuang ........ ------------------------------------------------------------------------ |