[Asterisk-video] amr audio continued

Mon Jul 14 15:00:15 CDT 2008

Hi Klaus,

I just tried ignoring the audio check and setting the
audioType,audioFormat,audioControl parameters anyways and then app_rtsp
will start getting the audio stream and I do hear audio in xlite.  It's
a bit delayed and the quality is awful, but I can make out enough to
confirm that it is the actual audio stream from the camera.  This also
means amr audio must be working in my asterisk install because the
camera sends amr and x-lite only has g711u enabled.

So apparently the check "if (sdp->audio->formats[i]->format &
chan->nativeformats)" shouldn't be failing.  The question becomes, is
the check not written properly, or is my install of asterisk not
configured properly or not setting the nativeformats parameter properly?
I installed asterisk using the directions from http://sip.fontventa.com/
to build it with amr audio support.  It's version 1.4.21

The value for sdp->audio->formats[i]->format is hex 0x2000 (8192 dec)
which is the proper value for amr so I don't think that is a problem.

My sip.conf enables all codecs and is very basic so it appears ok to me.
I am fairly new to asterisk and new to app_rtsp so I could be making a
total newb mistake.

Andrew

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Today's Topics:

   1. Re: can't hear amr audio (Klaus Darilion)

----------------------------------------------------------------------

Message: 1
Date: Mon, 14 Jul 2008 10:46:32 +0200
From: Klaus Darilion <klaus.mailinglists at pernau.at>
Subject: Re: [Asterisk-video] can't hear amr audio
To: Development discussion of video media support in Asterisk
	<asterisk-video at lists.digium.com>
Message-ID: <487B1268.80606 at pernau.at>
Content-Type: text/plain; charset=windows-1252; format=flowed

Which codec is: fmt 00002000 ? I think this is 2>>13 = AMR.

Thus I am irritated by " 0x580004 (ulaw|h263|h263p) to 0x0 (nothing)"

The 0x0 should be 0x2000 (AMR) - try to find out why the AMR bitmask 
flag is lost when calling set_format().

regards
klaus

Andrew Buchanan schrieb:
> Hi,
> 
>  
> 
> My camera (a Vivotek PZ7152) is set to AMR audio and when app_rtsp
connects it correctly identifies an audio track containing amr audio
AMR/8000, but I get no audio, only video.  I would appreciate some help
or suggestions to get audio working.
> 
>  
> 
> ------------------------
> 
> I compiled asterisk using the instructions to enable amr audio
> 
> http://sip.fontventa.com/svn/asterisk/amr/README
> 
>  
> 
> and I added the octet-aligned setting, which according to the sdp from

> the camera is the correct setting.
> 
> [amr]
> 
> octet-aligned=1
> 
>  
> 
> ------------------------
> 
>  
> 
> I?ve done some debugging in app_rtsp.c and found that:
> 
>  
> 
> This comparision ?if (sdp->audio->formats[i]->format &
chan->nativeformats)? in app_rtsp.c fails.
> 
>  
> 
> the value of format is 0x00002000
> 
> the value of nativeformats is 0x00580004
> 
>  
> 
> I also get this warning
> 
> [Jul 11 16:36:45] WARNING[31498]: channel.c:2800 set_format: Unable to
find a codec translation path from 0x580004 (ulaw|h263|h263p) to 0x0
(nothing)
> 
>  
> 
> I?m not sure why only ulaw,h263,h263p show up, but they are the only
protocols I have enabled in xlite, so that could be it, but the 0x0
(nothing) I don?t know.
> 
>  
> 
> Some debug code (my app_rtsp line #?s might be off a bit as I added
some debug statements)
> 
>  
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:421 RtspPlayerDescribe:
>DESCRIBE [/live2.sdp]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:442 RtspPlayerDescribe:
<DESCRIBE [/live2.sdp]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:1081 rtsp_play: -rtsp play
loop [0]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:1190 rtsp_play: -Receiving
describe
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line [v=0]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[o=RTSP 1215786675 889 IN IP4 0.0.0.0]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[s=RTSP server]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line [c=IN
IP4 0.0.0.0]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line [t=0
0]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=charset:Shift_JIS]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=range:npt=0-]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=control:*]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=etag:1234567890]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[m=video 0 RTP/AVP 96]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:689 CreateMedia: -creating
media [1,m=video 0 RTP/AVP 96]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[b=AS:0]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=rtpmap:96 MP4V-ES/30000]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=control:trackID=2]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=fmtp:96
profile-level-id=3;config=000001B003000001B2464D5F5047204D6F6465000001B5
09000001000000012000C48881F450A041E1463F;decode_buf=76800]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[m=audio 0 RTP/AVP 97]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:689 CreateMedia: -creating
media [1,m=audio 0 RTP/AVP 97]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[b=AS:13]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=rtpmap:97 AMR/8000]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=control:trackID=4]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=maxptime:200]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=fmtp:97 decode_buf=400;octet-align=1]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[1F450A041E1463F;decode_buf=76800]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[m=audio 0 RTP/AVP 97]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:689 CreateMedia: -creating
media [1,m=audio 0 RTP/AVP 97]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[b=AS:13]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=rtpmap:97 AMR/8000]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=control:trackID=4]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=maxptime:200]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:744 CreateSDP: -line
[a=fmtp:97 decode_buf=400;octet-align=1]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:1249 rtsp_play: -audio
[8192,97,trackID=4]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:1264 rtsp_play:
!(sdp->audio->formats[i]->format & chan->nativeformats) fmt 00002000
nfmt 00580004 -audio [97,trackID=4]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:1274 rtsp_play: -video
[4194304,96,trackID=2]
> 
> [Jul 11 16:36:45] WARNING[31498]: channel.c:2800 set_format: Unable to
find a codec translation path from 0x580004 (ulaw|h263|h263p) to 0x0
(nothing)
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:505 RtspPlayerSetupVideo:
-SETUP VIDEO [trackID=2]
> 
> [Jul 11 16:36:45] DEBUG[31498]: app_rtsp.c:557 RtspPlayerPlay: -PLAY
[/live2.sdp]
> 
> 
>
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