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
Attilla De Groot a ?crit :> 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/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
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=5162And 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.> AttillaDan