Re: [Asterisk-Users] Change g729 payload

2006-06-02 Thread Attilla De Groot


On Jun 1, 2006, at 10:57 PM, Dan Austin wrote:

It was put in a development branch, but has not seen any action in
over two months.

I have had it running against chan_ooh323 for six months and chan_sip
for three weeks in production for a moderately loaded conferencing
server.  Absolutely no issues with either channel.  It would not be
too hard to add support to any of the RTP-based channels, but IAX
will not work with the code as it stands.


Dan
__


Hi Dan,


That's very good news.

But to be honest, I'm not really a programmer and I'm not sure how to  
apply this patch on the source code. Could you give any instuctions  
about how to apply the patch ?



Attilla
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Re: [Asterisk-Users] Change g729 payload

2006-06-02 Thread Alejandro Vargas

2006/6/1, Attilla De Groot [EMAIL PROTECTED]:

Well I just found this:
http://bugs.digium.com/view.php?id=5162

So it seems that there is a patch and that it's ready for 2 months,
but I just checked the rtc.c code and it doesn't include this patch.
And I don't like to use beta patches on a production machine.


Download the source code for the version that you are using and patch
it yourself. It should not be so difficult.

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Alejandro Vargas
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Re: [Asterisk-Users] Change g729 payload

2006-06-02 Thread Alejandro Vargas

2006/6/2, Attilla De Groot [EMAIL PROTECTED]:

But to be honest, I'm not really a programmer and I'm not sure how to
apply this patch on the source code. Could you give any instuctions
about how to apply the patch ?


To apply a .patch file, you should use patch -pn, but if your source
code is much different than the version for what the patch was done,
it will not work. Then, you will need some knoledghe of programming.
You read the patch file, look what says the line to be modified and
what must it look like, then edit your source, find the corresponding
line (what may be similar to the one of the newer version for what the
patch has been done) and modify it with your favorite text editor (vi
rulez!). Then compile...

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Alejandro Vargas
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[Asterisk-Users] Change g729 payload

2006-06-01 Thread Attilla De Groot

Hi All,


I have a SIP provider that tells me that my RTP stream uses a  
20bytes payload in the g729 coded data. And they would like that we  
change this to 30bytes (3 frames).


But maybe I'm wrong but isn't a certain payload just a standard for a  
codec ?


And if I'm wrong, how can I change the payload for my g729 calls in   
Asterisk.



Greetings,
Attilla
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Re: [Asterisk-Users] Change g729 payload

2006-06-01 Thread Jean-Michel Hiver

Attilla De Groot a écrit :


Hi All,


I have a SIP provider that tells me that my RTP stream uses a  
20bytes payload in the g729 coded data. And they would like that we  
change this to 30bytes (3 frames).


But maybe I'm wrong but isn't a certain payload just a standard for a  
codec ?


You're wrong :)

And if I'm wrong, how can I change the payload for my g729 calls in   
Asterisk.


I had the same problem. Unfortunately this value is hard coded in 
Asterisk's code. I don't know if recent versions of Asterisk support this.


Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP  Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

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Re: [Asterisk-Users] Change g729 payload

2006-06-01 Thread Attilla De Groot

On Jun 1, 2006, at 5:38 PM, Jean-Michel Hiver wrote:



You're wrong :)


Nobody is perfect. ;)

I had the same problem. Unfortunately this value is hard coded in  
Asterisk's code. I don't know if recent versions of Asterisk  
support this.


Well I just found this:
http://bugs.digium.com/view.php?id=5162

So it seems that there is a patch and that it's ready for 2 months,  
but I just checked the rtc.c code and it doesn't include this patch.  
And I don't like to use beta patches on a production machine.



Attilla
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RE: [Asterisk-Users] Change g729 payload

2006-06-01 Thread Dan Austin
Attilla wrote:
 On Jun 1, 2006, at 5:38 PM, Jean-Michel Hiver wrote:

 You're wrong :)

 Nobody is perfect. ;)

 I had the same problem. Unfortunately this value is hard coded in  
 Asterisk's code. I don't know if recent versions of Asterisk  
 support this.

 Well I just found this:
 http://bugs.digium.com/view.php?id=5162
And I was just about to point that out...

 So it seems that there is a patch and that it's ready for 2 months,  
 but I just checked the rtc.c code and it doesn't include this patch.  
 And I don't like to use beta patches on a production machine.
It was put in a development branch, but has not seen any action in
over two months.

I have had it running against chan_ooh323 for six months and chan_sip
for three weeks in production for a moderately loaded conferencing 
server.  Absolutely no issues with either channel.  It would not be
too hard to add support to any of the RTP-based channels, but IAX 
will not work with the code as it stands.

 Attilla


Dan
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