[Asterisk-video] h324m and SIP

Fri Jan 11 05:04:58 CST 2008

Simple way?
Bigger video, smaller audio.. :)

BR
Sergio

----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
To: asterisk-video at lists.digium.com
Sent: Fri, 11 Jan 2008 11:57:42 +0100
Subject: Re: [Asterisk-video] h324m and SIP

Sergio Garcia Murillo schrieb:
> Have you tried to enable media dumps, rename to .h263 and .amr and play them with a media player??

How to enable media dumps? I have enabled h324m debug with "h324m debug 
level 9".

---Sr-x--T 1 root root  541760 2008-01-11 11:53 h223_in_825ea28.raw
---Sr-x--T 1 root root  541760 2008-01-11 11:53 h223_out_825ea28.raw
------S--T 1 root root 3588948 2008-01-11 11:53 h245_825e600.log
--w-r-S--- 1 root root 3588948 2008-01-11 11:53 h245_out_825e4b0.log
-r-Sr-x--- 1 root root  207868 2008-01-11 11:53 media_8245938.raw
-r----S--T 1 root root   82832 2008-01-11 11:53 media_8267700.raw

what is h263 and what is AMR?

thanks
klaus

> 
> BR
> Sergio
> 
> ----- Original Message -----
> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
> To: asterisk-video at lists.digium.com
> Sent: Fri, 11 Jan 2008 10:57:15 +0100
> Subject: Re: [Asterisk-video] h324m and SIP
> 
> 
> 
> Sergio Garcia Murillo schrieb:
>> Hi Klaus,
>>
>> Video from eyebean to 3G is never going to work directly, the bandwith it send is just too high, 
>> you should use app_transcoder to fix it.
> 
> With rev172 even SIP->3G video works fine.
> 
> How to use the app_transcoder? Can you give me an example which should work?
> 
>> I had no problems with amr conversion at all, is 3g to sip audio working fine?
> 
> Yes. 3g-->SIP audio is working fine.
> 
>> Could you try stopping video to see if audio get's better?
> 
> No difference.
> 
>> In the multiplexing h245 and audio should have priority over video.
> 
> Can I see somewhere in the log files if some buffer gets to big and 
> frames get  dropped?
> 
> thanks
> klaus
> 
>> BR
>> Sergio
>>
>> ----- Original Message -----
>> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
>> To: asterisk-video at lists.digium.com
>> Sent: Fri, 11 Jan 2008 10:22:51 +0100
>> Subject: [spam]Re: [Asterisk-video] h324m and SIP
>>
>> Thanks for all your input: Meanwhile I have manged to run Asterisk 
>> 1.4.17 and rev207.
>>
>> I test the h324m application bridging to SIP.
>>
>> My experiences:
>>
>> 3G-->h324m_gw---SIP+GSM--->Cisco-gw--->ISDN: audio is fine
>> 3G-->h324m_gw---SIP+GSM--->eyebeam: audio SIP->3G has drop outs
>> 3G-->h324m_gw---SIP+G711-->eyebeam: audio SIP->3G has many drop outs
>>
>> Here I suspect maybe some issues with bad conversion to AMR inside 
>> Asterisk. What results do you have?
>>
>> Video:
>> Video from 3G to SIP is working fine (eyebeam).
>> Video from SIP to 3G is bad - most of the picture is just black. Here I 
>> suspect maybe a problem if the bandwidth of the video received from SIP 
>> is to big to fit into the H223 channel.
>>
>> Sergio - how is multiplexing between Audio and Video handled - das Audio 
>> have fixed bandwidth or my too big video bandwidth also disturb audio?
>>
>> Are there somewhere bandwidth statistics in log files from h324m_gw or 
>> libh324m?
>>
>> thanks
>> klaus
>>
>> Klaus Darilion schrieb:
>>> Hi!
>>>
>>> Last time I tested h324m_gw with SIP clients audio and video worked fine 
>>> in both directions (xlite+nokia 6630). This was done with Asterisk 1.4.8 
>>> and fontventa rev163.
>>>
>>> Now I tried with Asterisk 1.4.17 and fontventa rev207 and audio does not 
>>> work and video works only from 3G to SIP.
>>>
>>> Using Asterisk 1.4.17 with fontventa rev163 makes video working fine 
>>> again, but Audio is still broken.
>>>
>>> Thus since Asterisk 1.4.8 something has changed that makes AMR 
>>> conversion broken and since fontventa rev. 163 something has changed 
>>> that makes video from SIP->3G broken.
>>>
>>> Now, I want to find out why current versions do not work. Thus, I would 
>>> be happy if you could tell me which version you use successfully to 
>>> track down the problem.
>>>
>>> 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
>> _______________________________________________
>> --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
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> --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