[Asterisk-Users] chan_h323: forcing 20ms packetisation

[David Hindmarsh]

HI Mike,

You wouldn't be trying to connect to Comindico in Australia by any
chance?



> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Mike O'Connor
> Sent: Monday, 18 October 2004 02:05
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] chan_h323: forcing 20ms packetisation
> 
> 
> Hi all
> 
> I spent a few hours trying to information on asterisk, h323 
> and sip support for codecs with 20ms packetisation, and have 
> not been able to find anything relivatant.
> 
> Our supplier of call termination requires h323 the following:
> 
> * The signalling port is 1720
> * H.323 version 2 with fast start and H.245 Tunneling.
> * The call should be initialised as Gateway-Gateway not using RAS.
> * The codecs supported are G.729, G.711alaw and G.711ulaw all 
> at 20 millisecond packetisation. Your equipment must support 
> all three and be able to dynamically negotiate these during 
> call setup.
> * We use RFC 2833 for out-of-band DTMF. Your equipment must 
> support this. The NTE RTP Payload type supported is 99.
> 
> I was able after reading the source code in chan_h323.c to 
> work out how to enable fast start and h.245 tunneling.
> 
> But the 20ms packetisation has me beat.
> 
> I have made a test call to the provider which did not work 
> becase I was sending 30ms voice packets.
> 
> SO my question does any one know now to force the correct 
> voice packet size ?
> 
> Thanks
> 
> Mike
> 
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