[Asterisk-video] asterisk-video Digest, Vol 28, Issue 16
Krishnakanth Chilakapati
krishnakanth.ch at tanlasolutions.com
Tue Aug 26 11:20:30 CDT 2008
We are getting the following warning when I try from 3G to SIP
[Aug 26 16:59:15] WARNING[27680]: chan_sip.c:3709 sip_write: Asked to transmit frame type 8192, while native formats is 0x8 (alaw)(8) read/write = 0x0 (nothing)(0)/0x0 (nothing)(0)
[Aug 26 16:59:15] WARNING[27680]: chan_sip.c:3709 sip_write: Asked to transmit frame type 8192, while native formats is 0x8 (alaw)(8) read/write = 0x0 (nothing)(0)/0x0 (nothing)(0)
[Aug 26 16:59:15] WARNING[27680]: chan_sip.c:3709 sip_write: Asked to transmit frame type 8192, while native formats is 0x8 (alaw)(8) read/write = 0x0 (nothing)(0)/0x0 (nothing)(0)
[Aug 26 16:59:15] WARNING[27680]: chan_sip.c:3709 sip_write: Asked to transmit frame type 8192, while native formats is 0x8 (alaw)(8) read/write = 0x0 (nothing)(0)/0x0 (nothing)(0)
[Aug 26 16:59:15] WARNING[27680]: chan_sip.c:3709 sip_write: Asked to transmit frame type 8192, while native formats is 0x8 (alaw)(8) read/write = 0x0 (nothing)(0)/0x0 (nothing)(0)
Krishna
-----Original Message-----
From: asterisk-video-bounces at lists.digium.com [mailto:asterisk-video-bounces at lists.digium.com] On Behalf Of asterisk-video-request at lists.digium.com
Sent: Tuesday, August 26, 2008 9:05 PM
To: asterisk-video at lists.digium.com
Subject: asterisk-video Digest, Vol 28, Issue 16
Send asterisk-video mailing list submissions to
asterisk-video at lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-video
or, via email, send a message with subject or body 'help' to
asterisk-video-request at lists.digium.com
You can reach the person managing the list at
asterisk-video-owner at lists.digium.com
When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-video digest..."
Today's Topics:
1. Re: 3G video delay (Sergio Garcia Murillo)
2. Re: 3G video delay (Dan Julius)
3. Re: 3G video delay (Emmanuel BUU)
4. Re: 3G video delay (Sergio Garcia Murillo)
5. Re: version of gnash/gstreamer/ffmpeg to be usedwithapp_swf
(Sergio Garcia Murillo)
6. Re: asterisk-video Digest, Vol 28, Issue 15
(Krishnakanth Chilakapati)
----------------------------------------------------------------------
Message: 1
Date: Mon, 25 Aug 2008 19:58:01 +0200
From: "Sergio Garcia Murillo" <sergio.garcia at fontventa.com>
Subject: Re: [Asterisk-video] 3G video delay
To: <asterisk-video at lists.digium.com>
Message-ID: <000A5DCCB33C4DD2BF733FAE74B86DA2.MAI at fontventa.es>
Content-Type: text/plain; charset="iso-8859-1"
Hi Klaus,
Modify H223AL2Sender::SendPDU() in H324MAL2.cpp
Check if jitBuffer.GetSize()>YOURVALUE and return
BR
Sergio
----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
To: asterisk-video at lists.digium.com
Sent: Mon, 25 Aug 2008 14:49:18 +0200
Subject: [Asterisk-video] 3G video delay
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
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
------------------------------
Message: 2
Date: Mon, 25 Aug 2008 21:14:26 +0300
From: "Dan Julius" <dan.julius at gmail.com>
Subject: Re: [Asterisk-video] 3G video delay
To: "Development discussion of video media support in Asterisk"
<asterisk-video at lists.digium.com>
Message-ID:
<131b8bd40808251114h7fc03854rf53cabd74a9c2c9b at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
I was just wondering if it is safe to just drop frames?
How do you know if you are not dropping part of an I frame, which will later
invalidate all following P frames?
Thanks,
Dan
On Mon, Aug 25, 2008 at 8:58 PM, Sergio Garcia Murillo <
sergio.garcia at fontventa.com> wrote:
> Hi Klaus,
>
> Modify H223AL2Sender::SendPDU() in H324MAL2.cpp
> Check if jitBuffer.GetSize()>YOURVALUE and return
>
> BR
> Sergio
>
> ----- Original Message -----
> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
> To: asterisk-video at lists.digium.com
> Sent: Mon, 25 Aug 2008 14:49:18 +0200
> Subject: [Asterisk-video] 3G video delay
>
> 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
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-video/attachments/20080825/580da5b9/attachment-0001.htm
------------------------------
Message: 3
Date: Mon, 25 Aug 2008 21:02:09 +0200
From: Emmanuel BUU <emmanuel.buu at ives.fr>
Subject: Re: [Asterisk-video] 3G video delay
To: Development discussion of video media support in Asterisk
<asterisk-video at lists.digium.com>
Message-ID: <48B301B1.8010801 at ives.fr>
Content-Type: text/plain; charset="iso-8859-1"
Dan Julius a ?crit :
> Hi,
>
> I was just wondering if it is safe to just drop frames?
> How do you know if you are not dropping part of an I frame, which will
> later invalidate all following P frames?
The only way to know is to dig into the headers of the H.263 payload.
One remark also: what should be dropped is not only an I frame but an I
frame and all the subsquent P frames up to the next I frame.
I fear that such an algorithm would lead to a very bad quality.
Transcoding is the only way to go.
Emmanuel
>
> Thanks,
> Dan
>
> On Mon, Aug 25, 2008 at 8:58 PM, Sergio Garcia Murillo
> <sergio.garcia at fontventa.com <mailto:sergio.garcia at fontventa.com>> wrote:
>
> Hi Klaus,
>
> Modify H223AL2Sender::SendPDU() in H324MAL2.cpp
> Check if jitBuffer.GetSize()>YOURVALUE and return
>
> BR
> Sergio
>
> ----- Original Message -----
> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at
> <mailto:klaus.mailinglists at pernau.at>]
> To: asterisk-video at lists.digium.com
> <mailto:asterisk-video at lists.digium.com>
> Sent: Mon, 25 Aug 2008 14:49:18 +0200
> Subject: [Asterisk-video] 3G video delay
>
> 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
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-video/attachments/20080825/41f28b37/attachment-0001.htm
------------------------------
Message: 4
Date: Mon, 25 Aug 2008 22:10:24 +0200
From: "Sergio Garcia Murillo" <sergio.garcia at fontventa.com>
Subject: Re: [Asterisk-video] 3G video delay
To: <asterisk-video at lists.digium.com>
Message-ID: <17FC34D2CA7C4A9E8F6A5668AEB890C2.MAI at fontventa.es>
Content-Type: text/plain; charset="iso-8859-1"
Of course, transcoding with my app_transcoder is the best option.. jejeje ;)
It could be possible also to send a fast update video (if the client supports it) when you reset the video channel in order to minimize the problem.
Anyway let us know the tests results.
BR
Sergio
----- Original Message -----
From: Emmanuel BUU [mailto:emmanuel.buu at ives.fr]
To: asterisk-video at lists.digium.com
Sent: Mon, 25 Aug 2008 21:02:09 +0200
Subject: Re: [Asterisk-video] 3G video delay
Dan Julius a ?crit :
> Hi,
>
> I was just wondering if it is safe to just drop frames?
> How do you know if you are not dropping part of an I frame, which will
> later invalidate all following P frames?
The only way to know is to dig into the headers of the H.263 payload.
One remark also: what should be dropped is not only an I frame but an I
frame and all the subsquent P frames up to the next I frame.
I fear that such an algorithm would lead to a very bad quality.
Transcoding is the only way to go.
Emmanuel
>
> Thanks,
> Dan
>
> On Mon, Aug 25, 2008 at 8:58 PM, Sergio Garcia Murillo
> <sergio.garcia at fontventa.com <mailto:sergio.garcia at fontventa.com>> wrote:
>
> Hi Klaus,
>
> Modify H223AL2Sender::SendPDU() in H324MAL2.cpp
> Check if jitBuffer.GetSize()>YOURVALUE and return
>
> BR
> Sergio
>
> ----- Original Message -----
> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at
> <mailto:klaus.mailinglists at pernau.at>]
> To: asterisk-video at lists.digium.com
> <mailto:asterisk-video at lists.digium.com>
> Sent: Mon, 25 Aug 2008 14:49:18 +0200
> Subject: [Asterisk-video] 3G video delay
>
> 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
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
------------------------------
Message: 5
Date: Tue, 26 Aug 2008 11:10:09 +0200
From: "Sergio Garcia Murillo" <sergio.garcia at fontventa.com>
Subject: Re: [Asterisk-video] version of gnash/gstreamer/ffmpeg to be
usedwithapp_swf
To: <asterisk-video at lists.digium.com>
Message-ID: <FFBF71921936455B9E3E3DC940CF8BEF.MAI at fontventa.es>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
I've upgraded the flash support, now it supports gnash 0.8.3 (you should have to compile it with --enable-cassert=no) and I have fixed some video problems.
Feedbacks are welcome.
BR
Sergio
----- Original Message -----
From: Sergio Garcia Murillo [mailto:sergio.garcia at fontventa.com]
To: asterisk-video at lists.digium.com
Sent: Fri, 22 Aug 2008 09:54:14 +0200
Subject: Re: [Asterisk-video] version of gnash/gstreamer/ffmpeg to be usedwithapp_swf
Hi Low,
When I developed the swf support, gnash didn't have any stable release yet. I received a patch a while ago for compiling it against gnash 8.2, I was checking it before going on holidays, so I'll restart the work now.
FFmpeg version is always the latest svn version (or almost) and I don't use gstreamer at all (check the version needed by gnash 8.2).
Best regards
Sergio
----- Original Message -----
From: Low Yu Siang [mailto:yusiang at yahoo.com]
To: asterisk-video at lists.digium.com
Sent: Thu, 21 Aug 2008 04:17:42 -0700 (PDT)
Subject: [Asterisk-video] version of gnash/gstreamer/ffmpeg to be used withapp_swf
Hi!
Has anyone tried out sergio's experimental app_swf? May I know which version/SVN revision of gnash, gstreamer and ffmpeg that you are using together with app_swf?
Regards,
Low Yu Siang
Send instant messages to your online friends http://uk.messenger.yahoo.com
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
------------------------------
Message: 6
Date: Tue, 26 Aug 2008 21:09:20 +0530
From: Krishnakanth Chilakapati <krishnakanth.ch at tanlasolutions.com>
Subject: Re: [Asterisk-video] asterisk-video Digest, Vol 28, Issue 15
To: "asterisk-video at lists.digium.com"
<asterisk-video at lists.digium.com>
Message-ID:
<53A591B34ED06740BCB21D03F6C5748D0863D8AABA at pegasus.tanlasolutions.com>
Content-Type: text/plain; charset="us-ascii"
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-----
From: asterisk-video-bounces at lists.digium.com [mailto:asterisk-video-bounces at lists.digium.com] On Behalf Of asterisk-video-request at lists.digium.com
Sent: Monday, August 25, 2008 10:30 PM
To: asterisk-video at lists.digium.com
Subject: asterisk-video Digest, Vol 28, Issue 15
Send asterisk-video mailing list submissions to
asterisk-video at lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-video
or, via email, send a message with subject or body 'help' to
asterisk-video-request at lists.digium.com
You can reach the person managing the list at
asterisk-video-owner at lists.digium.com
When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-video digest..."
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
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
------------------------------
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
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-video/attachments/20080825/fc74eb33/attachment.html
------------------------------
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
------------------------------
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
End of asterisk-video Digest, Vol 28, Issue 15
**********************************************
------------------------------
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
End of asterisk-video Digest, Vol 28, Issue 16
**********************************************
More information about the asterisk-video
mailing list