[Asterisk-video] asterisk-video Digest, Vol 28, Issue 16
Klaus Darilion
klaus.mailinglists at pernau.at
Tue Aug 26 17:40:26 CDT 2008
Do you have allowed AMR codec in sip.conf?
allow=all
regards
klaus
Krishnakanth Chilakapati wrote:
> 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
>
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> 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
>>
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> ------------------------------
>
> 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
>
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> ------------------------------
>
> 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
>>>
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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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