[Asterisk-video] Help with rtsp

Tue Jun 24 07:46:12 CDT 2008

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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