Posted: Fri Jan 11, 2008 1:26 pm Post subject: [Asterisk-video] h324m and SIP
Yes you're rigth I think only incoming media is dumped to a file.
I use object pointer to create the file name, it was the quickest way, I'll try to improve it in the future.
BR
Sergio
----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
To: asterisk-video@lists.digium.com
Sent: Fri, 11 Jan 2008 12:30:19 +0100
Subject: Re: [Asterisk-video] h324m and SIP
Sergio Garcia Murillo schrieb:
Quote:
Simple way?
Bigger video, smaller audio.. :)
Shouldn't there be 4 media files? for both directions?
Quote:
How to enable media dumps? I have enabled h324m debug with "h324m debug
level 9".
> BR
> Sergio
>
> ----- Original Message -----
> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
> To: asterisk-video@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@pernau.at]
>> To: asterisk-video@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
>> _______________________________________________
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>> asterisk-video mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-video
>>
>>
>>
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>>
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>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-video
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Posted: Fri Jan 11, 2008 1:29 pm Post subject: [Asterisk-video] h324m and SIP
Klaus Darilion schrieb:
Quote:
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?
Fixed: I had to disable "acoustic echo cancellation" in eyebeam (I had
physical echo as the headset and the 3G handset were in the same room)
Quote:
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.
Fixed: I had to comment ResetMedia queue in h324m_gw
regards
klaus
Quote:
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
_______________________________________________
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Posted: Fri Jan 11, 2008 2:16 pm Post subject: [Asterisk-video] h324m and SIP
Hi Reza!
IIRC you had to patch libpri to send low layer compatibility to signal
3G calls on ISDN. Is this patch public available? I need to use low
layer compatibility too.
thanks
klaus
Reza Fatahillah schrieb:
Quote:
I'm using 1.4.9 BRI version with fonventa rev 207
and for PRI i used 1.4.13 with fontventa rev 199(or
above forget :) not using it anymore)
--- Klaus Darilion <klaus.mailinglists@pernau.at>
wrote:
> 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.
____________________________________________________________________________________
Never miss a thing. Make Yahoo your home page.
http://www.yahoo.com/r/hs
_______________________________________________
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Posted: Wed Jan 16, 2008 1:06 pm Post subject: [Asterisk-video] h324m and SIP
Sergio Garcia Murillo schrieb:
Quote:
Have you tried to enable media dumps, rename to .h263 and .amr and play them with a media player??
Hi Sergio:
I renamed the larger file to .263. VLC and mplayer can playback it, but
too fast. Using mp4creator I can create an mp4 file with correct
duration (frame rate). (using ffmpeg it also uses a wrong framerate and
video is too fast).
Do you know how mp4creater knows the proper frame rate but ffmpeg fails?
Further, I renamed the small file to .amr, but no application can
process it (VLC, mplayer, ffmpeg. mpeg4ip). Thus, are you sure this is a
valid AMR file?
regards
klaus
Quote:
BR
Sergio
----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
To: asterisk-video@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@pernau.at]
> To: asterisk-video@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
_______________________________________________
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Posted: Wed Jan 16, 2008 1:42 pm Post subject: [Asterisk-video] h324m and SIP
Hi Klaus,
A raw h263 stream file has no timing information, only a time reference that is scaled by the base reference time.
In fact when creating an mp4 file you should specify the fps when adding a h263 track to get correct playback speed.
To create a valid amr file you have to prepend a 6 bytes header to the stream:
----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
To: asterisk-video@lists.digium.com
Sent: Wed, 16 Jan 2008 14:00:18 +0100
Subject: Re: [Asterisk-video] h324m and SIP
Sergio Garcia Murillo schrieb:
Quote:
Have you tried to enable media dumps, rename to .h263 and .amr and play them with a media player??
Hi Sergio:
I renamed the larger file to .263. VLC and mplayer can playback it, but
too fast. Using mp4creator I can create an mp4 file with correct
duration (frame rate). (using ffmpeg it also uses a wrong framerate and
video is too fast).
Do you know how mp4creater knows the proper frame rate but ffmpeg fails?
Further, I renamed the small file to .amr, but no application can
process it (VLC, mplayer, ffmpeg. mpeg4ip). Thus, are you sure this is a
valid AMR file?
regards
klaus
Quote:
BR
Sergio
----- Original Message -----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
To: asterisk-video@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@pernau.at]
> To: asterisk-video@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
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
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>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
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