[Asterisk-Users] chan_h323: forcing 20ms packetisation

[Mike O'Connor]

Hi All

Is there a better mailing list where I should ask these questions ?

Thanks

Mike O'Connor wrote:

> 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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