[Asterisk-Users] chan_h323: forcing 20ms packetisation

[Michael M. Saunders]

Is this mike oconnor as in the Australian mick oconnor

-----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: Wednesday, 20 October 2004 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] chan_h323: forcing 20ms packetisation

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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