[Asterisk-video] Help with rtsp

Tue Jun 24 15:28:51 CDT 2008

I test with a softphone, a videophone and a 3g phone. The only
difference is with the 3g phone the message is:

 [Jun 24 11:38:54] WARNING[13648]: channel.c:2781 set_format: Unable to
> find a codec translation path from unknown to unknown

2008/6/24 Sergio Garcia Murillo <sergio.garcia at fontventa.com>:
> What are u using as the other end point? a softphone or a 3g phone with app_h324m?
>
> BR
> Sergio
> ----- Original Message -----
> From: Jesús Gumiel [mailto:jesus.gumiel at gmail.com]
> To: asterisk-video at lists.digium.com
> Sent: Tue, 24 Jun 2008 11:38:22 +0200
> Subject: Re: [Asterisk-video] Help with rtsp
>
> Hi Sergio,
>
>  I tried to execute app_rtsp without app_transcode, and the result is the next:
>
>    -- Executing [8000 at default:1] Answer("SIP/5004-158c2920", "") in new stack
>    -- Executing [8000 at default:2] rtsp("SIP/5004-158c2920",
> "rtsp://80.37.220.217:8001/sample_100kbit.mp4") in new stack
> [Jun 24 11:38:54] WARNING[13648]: app_rtsp.c:1033 rtsp_play: >rtsp play
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:286 GetUdpPorts:
> -GetUdpPorts [35224,35225]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:286 GetUdpPorts:
> -GetUdpPorts [35226,35227]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:422 RtspPlayerDescribe:
>>DESCRIBE [/sample_100kbit.mp4]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:443 RtspPlayerDescribe:
> <DESCRIBE [/sample_100kbit.mp4]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:1082 rtsp_play: -rtsp play loop
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:1161 rtsp_play: -Receiving describe
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:1161 rtsp_play: -Receiving describe
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line [A283020F2A21F]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=mpeg4-esid:201]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [m=audio 0 RTP/AVP 97]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:690 CreateMedia: -creating
> media [1,m=audio 0 RTP/AVP 97]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line [b=AS:20]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=rtpmap:97 mpeg4-generic/8000/2]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=control:trackID=4]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=fmtp:97 profile-level-id=15;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3;config=1590]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=mpeg4-esid:101]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [GxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBWUVDQVV3Rkh3TklBTWtnQUdVRUx5QVJBRzNkQUFFaytBQUJKUGdGSUFBQUFiRHpBQUFCdFE3Z1FNRFBBQUFCQUFBQUFTQUFoRUQ2S0RBZzhxSWZCaEFBUkFBQUFsZ0FBQUFBSUFBQUFBQURBVElDbndNdUFHVUFCSUNBZ0JSQUZRQVlBQUFBVGlBQUFFNGdCWUNBZ0FJVmtBWVFBRVFBQUI5QUFBQWZRQ0FnQUFBQUF3PT0EDQEFAADIAAAAAAAAAAAGCQEAAAAAAAAAAANpAAJARmRhdGE6YXBwbGljYXRpb24vbXBlZzQtYmlmcy1hdTtiYXNlNjQsd0JBU2daTUNvRmNtRUVIOEFBQUIvQUFBQkVLQ0tDbjQEEgINAABkAAAAAAAAAAAFAwAAYAYJAQAAAAAAAAAA"]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=isma-compliance:1,1.0,1]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=range:npt=0-  70.00000]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [m=video 0 RTP/AVP 96]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:690 CreateMedia: -creating
> media [1,m=video 0 RTP/AVP 96]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line [b=AS:76]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=rtpmap:96 MP4V-ES/90000]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=control:trackID=3]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=cliprect:0,0,242,192]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=framesize:96 192-242]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=fmtp:96 profile-level-id=1;config=000001B0F3000001B50EE040C0CF0000010000000120008440FAAAAAAAAGCQEAAAAAAAAAAANpAAJARmRhdGE6YXBwbGljYXRpb24vbXBlZzQtYmlmcy1hdTtiYXNlNjQsd0JBU2daTUNvRmNtRUVIOEFBQUIvQUFBQkVLQ0tDbjQEEgINAABkAAAAAAAAAAAFAwAAYAYJAQAAAAAAAAAA"]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=isma-compliance:1,1.0,1]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:745 CreateSDP: -line
> [a=range:npt=0-  70.00000]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:1220 rtsp_play: -audio
> [0,-1,trackID=4]
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:1241 rtsp_play: -video
> [4194304,96,trackID=3]
> [Jun 24 11:38:54] WARNING[13648]: channel.c:2781 set_format: Unable to
> find a codec translation path from ulaw to unknown
> [Jun 24 11:38:54] ERROR[13648]: app_rtsp.c:1269 rtsp_play: No media found
> [Jun 24 11:38:54] DEBUG[13648]: app_rtsp.c:1506 rtsp_play: -rtsp_play
> end loop [0]
> [Jun 24 11:38:54] WARNING[13648]: app_rtsp.c:1532 rtsp_play: <rtsp_pla
>   -- Executing [8000 at default:3] Hangup("SIP/5004-158c2920", "") in
> new stack
>
> I don´t understand how I can use app_transcode alone, can you explain me??.
>
> Thanks in advance
>
>
> 2008/6/23 Sergio Garcia Murillo <sergio.garcia at fontventa.com>:
>> Why are you using app_transcoder with app_rtsp? can you try them independently first?
>>
>> BR
>> Sergio
>>
>> ----- Original Message -----
>> From: Jesús Gumiel [mailto:jesus.gumiel at gmail.com]
>> To: asterisk-video at lists.digium.com
>> Sent: Thu, 19 Jun 2008 14:42:35 +0200
>> Subject: [Asterisk-video] Help with rtsp
>>
>> I test app_rtsp with the video:
>> http://sip.fontventa.com/files/sample_300kbit_ulaw.3gp
>>
>> and I have the next errors:
>>
>> [Jun 19 14:42:15] DEBUG[26496]: app_transcoder.c:1074
>> av_log_asterisk_callback: [h263 @ 0x2aaab5cf6b60] header damaged
>> [Jun 19 14:42:15] DEBUG[26496]: app_transcoder.c:1074
>> av_log_asterisk_callback: [h263 @ 0x2aaab5cf6b60] Bad picture start
>> code
>> [Jun 19 14:42:15] DEBUG[26496]: app_transcoder.c:1074
>> av_log_asterisk_callback: [h263 @ 0x2aaab5cf6b60] header damaged
>> [Jun 19 14:42:15] DEBUG[26497]: app_transcoder.c:1074
>> av_log_asterisk_callback: [h263 @ 0x2aaab5cf6b60] vbv buffer overflow
>> [Jun 19 14:42:15] DEBUG[26497]: app_transcoder.c:1074
>> av_log_asterisk_callback: [h263 @ 0x2aaab5cf6b60] vbv buffer overflow
>> [Jun 19 14:42:15] DEBUG[26497]: app_transcoder.c:1074
>> av_log_asterisk_callback: [h263 @ 0x2aaab5cf6b60] vbv buffer overflow
>>
>> I hear the audio but i can´t see the video. Any idea?
>>
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>
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