[Asterisk-video] asterisk-video Digest, Vol 28, Issue 15
Krishnakanth Chilakapati
krishnakanth.ch at tanlasolutions.com
Tue Aug 26 10:39:20 CDT 2008
We are not able to open http://sip.fontventa.com/svn/asterisk/amr/ to download the latest patch. When we try to build using the instructions its not updating make files
Krishna
-----Original Message-----
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Sent: Monday, August 25, 2008 10:30 PM
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Subject: asterisk-video Digest, Vol 28, Issue 15
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Today's Topics:
1. Re: 3G <-->SIP audio problems (Klaus Darilion)
2. Re: Re :Re: 3G <-->SIP audio problems (Klaus Darilion)
3. app_transcoder (Klaus Darilion)
4. Re: app_transcoder (Sergio Garcia Murillo)
5. Re: 3G <-->SIP audio problems (aster vdo)
6. 3G video delay (Klaus Darilion)
----------------------------------------------------------------------
Message: 1
Date: Mon, 25 Aug 2008 10:45:15 +0200
From: Klaus Darilion <klaus.mailinglists at pernau.at>
Subject: Re: [Asterisk-video] 3G <-->SIP audio problems
To: aster vdo <astervdo at gmail.com>
Cc: asterisk-video at lists.digium.com
Message-ID: <48B2711B.205 at pernau.at>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Have you patched Asterisk with the AMR patch from sip.fontventa.com?
klaus
aster vdo schrieb:
> Hi,
>
> I am doing a video call from a 3G to SIP.
>
> The video works fine, but there is no audio for both the parties.
>
> I using Asterisk 1.4.20.1 <http://1.4.20.1>
>
> and i get the following warning message on asterisk
>
> WARNING[15840] chan_sip.c: Asked to transmit frame type 8192, while
> native formats is 0x4 (ulaw)(4) read/write = 0x0 (nothing)(0)/0x0
> (nothing)(0)
>
> and the show channel commands results in the following output.
>
>
> *CLI> core show channel Zap/1-1
> -- General --*CLI>
> Name: Zap/1-1
> Type: Zap
> UniqueID: 1219243635.23
> Caller ID: XXXXXXXX
> Caller ID Name: (N/A)
> DNID Digits: XXXXXXX
> State: Up (6)
> Rings: 1
> NativeFormats: 0x44 (ulaw|slin)
> WriteFormat: 0x4 (ulaw)
> ReadFormat: 0x4 (ulaw)
> WriteTranscode: No
> ReadTranscode: No
> 1st File Descriptor: 19
> Frames in: 3275
> Frames out: 2532
> Time to Hangup: 0
> Elapsed Time: 0h1m5s
> Direct Bridge: <none>
> Indirect Bridge: <none>
> -- PBX --
> Context: default
> Extension: s
> Priority: 2
> Call Group: 0
> Pickup Group: 0
> Application: h324m_gw
> Data: dial at cell_to_sip
> Blocking in: ast_waitfor_nandfds
> Variables:
> ul1=65535
> CALLEDTON=33
> ANI2=0
> TRANSFERCAPABILITY=DIGITAL
>
> CDR Variables:LI>
> level 1: clid=XXXXXXXX
> level 1: src=XXXXXXXX
> level 1: dst=s
> level 1: dcontext=default
> level 1: channel=Zap/1-1
> level 1: lastapp=h324m_gw
> level 1: lastdata=dial at cell_to_sip
> level 1: start=2008-08-20 15:47:15
> level 1: answer=2008-08-20 15:47:30
> level 1: end=2008-08-20 15:47:30
> level 1: duration=0
> level 1: billsec=0
> level 1: disposition=ANSWERED
> level 1: amaflags=DOCUMENTATION
> level 1: uniqueid=1219243635.23
>
>
>
> Is there any thing i am doing wrong..
>
>
> regards
>
> aster
------------------------------
Message: 2
Date: Mon, 25 Aug 2008 10:45:42 +0200
From: Klaus Darilion <klaus.mailinglists at pernau.at>
Subject: Re: [Asterisk-video] Re :Re: 3G <-->SIP audio problems
To: Development discussion of video media support in Asterisk
<asterisk-video at lists.digium.com>
Message-ID: <48B27136.6040205 at pernau.at>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
aster vdo schrieb:
> Hi Sergio,
>
> Thanks for the reply.
>
> I had installed the latest amr codec availabel from the following link
>
> http://sip.fontventa.com/svn/asterisk/amr
>
>
> let me know if it is the correct codec of amr to be used with asterisk..
yes it is the correct
------------------------------
Message: 3
Date: Mon, 25 Aug 2008 11:54:08 +0200
From: Klaus Darilion <klaus.mailinglists at pernau.at>
Subject: [Asterisk-video] app_transcoder
To: Development discussion of video media support in Asterisk
<asterisk-video at lists.digium.com>
Message-ID: <48B28140.4030508 at pernau.at>
Content-Type: text/plain; charset=ISO-8859-15; format=flowed
Hi!
From the fontventa website: "Currently only MPEG4-ES to H263 is
supported, H263 to H263 would be really soon."
I wonder is this still the case? Can't I use it to transcode H.263 to H.263?
Further, with the recent fixes to app_transcoder - is it now stable?
thanks
klaus
------------------------------
Message: 4
Date: Mon, 25 Aug 2008 12:24:49 +0200
From: "Sergio Garcia Murillo" <sergio.garcia at fontventa.com>
Subject: Re: [Asterisk-video] app_transcoder
To: <asterisk-video at lists.digium.com>
Message-ID: <1B6C465781C94C46962A3897DED9C992.MAI at fontventa.es>
Content-Type: text/plain; charset="iso-8859-1"
Hi Klaus
The input codecs are MPEG4-ES, H263-1996 and H263-1998/2000 and only H263-1998/2000 output codec.
The only issue left is the channel lock by sending in another thread, but I haven't experienced any trouble due to it in my tests.
Best regards
Sergio
----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
To: asterisk-video at lists.digium.com
Sent: Mon, 25 Aug 2008 11:54:08 +0200
Subject: [Asterisk-video] app_transcoder
Hi!
From the fontventa website: "Currently only MPEG4-ES to H263 is
supported, H263 to H263 would be really soon."
I wonder is this still the case? Can't I use it to transcode H.263 to H.263?
Further, with the recent fixes to app_transcoder - is it now stable?
thanks
klaus
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------------------------------
Message: 5
Date: Mon, 25 Aug 2008 16:06:30 +0530
From: "aster vdo" <astervdo at gmail.com>
Subject: Re: [Asterisk-video] 3G <-->SIP audio problems
To: "Klaus Darilion" <klaus.mailinglists at pernau.at>
Cc: asterisk-video at lists.digium.com
Message-ID:
<302c6de30808250336l75deec30j9caa2267131a7882 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi Klaus,
I had done the patch from sip.fontventa.com.
and also added the line
[amr]
octet-aligned=1
in my codecs.conf file.
but that did not help me.
regards,
aster
On 8/25/08, Klaus Darilion <klaus.mailinglists at pernau.at> wrote:
>
> Have you patched Asterisk with the AMR patch from sip.fontventa.com?
>
> klaus
>
> aster vdo schrieb:
>
>> Hi,
>>
>> I am doing a video call from a 3G to SIP.
>>
>> The video works fine, but there is no audio for both the parties.
>>
>> I using Asterisk 1.4.20.1 <http://1.4.20.1>
>>
>> and i get the following warning message on asterisk
>>
>> WARNING[15840] chan_sip.c: Asked to transmit frame type 8192, while native
>> formats is 0x4 (ulaw)(4) read/write = 0x0 (nothing)(0)/0x0 (nothing)(0)
>>
>> and the show channel commands results in the following output.
>>
>>
>> *CLI> core show channel Zap/1-1
>> -- General --*CLI>
>> Name: Zap/1-1
>> Type: Zap
>> UniqueID: 1219243635.23
>> Caller ID: XXXXXXXX
>> Caller ID Name: (N/A)
>> DNID Digits: XXXXXXX
>> State: Up (6)
>> Rings: 1
>> NativeFormats: 0x44 (ulaw|slin)
>> WriteFormat: 0x4 (ulaw)
>> ReadFormat: 0x4 (ulaw)
>> WriteTranscode: No
>> ReadTranscode: No
>> 1st File Descriptor: 19
>> Frames in: 3275
>> Frames out: 2532
>> Time to Hangup: 0
>> Elapsed Time: 0h1m5s
>> Direct Bridge: <none>
>> Indirect Bridge: <none>
>> -- PBX --
>> Context: default
>> Extension: s
>> Priority: 2
>> Call Group: 0
>> Pickup Group: 0
>> Application: h324m_gw
>> Data: dial at cell_to_sip
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> ul1=65535
>> CALLEDTON=33
>> ANI2=0
>> TRANSFERCAPABILITY=DIGITAL
>>
>> CDR Variables:LI>
>> level 1: clid=XXXXXXXX
>> level 1: src=XXXXXXXX
>> level 1: dst=s
>> level 1: dcontext=default
>> level 1: channel=Zap/1-1
>> level 1: lastapp=h324m_gw
>> level 1: lastdata=dial at cell_to_sip
>> level 1: start=2008-08-20 15:47:15
>> level 1: answer=2008-08-20 15:47:30
>> level 1: end=2008-08-20 15:47:30
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=ANSWERED
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1219243635.23
>>
>>
>>
>> Is there any thing i am doing wrong..
>>
>>
>> regards
>>
>> aster
>>
>
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Message: 6
Date: Mon, 25 Aug 2008 14:49:18 +0200
From: Klaus Darilion <klaus.mailinglists at pernau.at>
Subject: [Asterisk-video] 3G video delay
To: Development discussion of video media support in Asterisk
<asterisk-video at lists.digium.com>
Message-ID: <48B2AA4E.4070808 at pernau.at>
Content-Type: text/plain; charset=ISO-8859-15; format=flowed
Hi!
I have a problem with a SIP client which sends video with high bitrate
if there is very much movement. This adds huge delay to the video as it
does not fit anymore into the 64kbit channel.
As in normal scenarios the delay is fine I want to avoid transcoding.
Thus my idea was to drop video frames if the queue is getting to big.
I want to test may idea, but the question is - where to do this? I think
this have to be done in libh324m. Sergio, can you please tell me
how/where I could implement such a behavior?
regards
klaus
------------------------------
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