[asterisk-dev] Packetization update (bugid 5162)

[Dan Austin]

>>Testing feedback is welcomed.  A recent SVN checkout with the latest
>>patch in bugid 5162 will get you packetization core and implimentation
>>in chan_sip and chan_skinny.  If you can and want to test chan_ooh323,
>>then you'll also need a svn checkout from asterisk-addons and the
latest
>>patch in bugid 5588.
>>
>>Configuration is simply-
>>allow=codec:ms
>>	in chan_sip in [general] or per user/peer/friend
>>	in chan_skinny in [general] or per device
>>	in chan_ooh323 in [general] or per user/peer/friend
>>  
>>
>OK Dan. I'll try to get this built next week (or over the weekend if I 
>have the new servers i've ordered early) and I'll let you know how it
goes.

>NB: So you can have:

>allow=g729:20
>allow=g729:40
>allow=g729:60
>allow=g729:80

On four different users/peers.  The normal approach would be to
set the prefered packetization.  Note that Asterisk has happily
handled receiving audio packetization >20ms for a long time, it
just always used 20ms for transmitting.

An additional approach in chan_sip is to allow the channel to
identify the remote endpoints prefered packetization based
on the ptime element in the SDP, and using it.

So two approachs that this patch enables-
	1.  You know all of your endpoints and network's capabilities,
		so you can tune for latency and bandwidth.
	2.  You do not have control over all endpoints, but want to
		honor the ptime/packetization that any endpoint might
		request.

Dan

Say?

Cheers,
Jean-Michel.
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