[asterisk-dev] synchronizing RTP

[Kevin P. Fleming]

Klaus Darilion wrote:

> I have a problem with generating a nice RTP stream with one packet every 
> 20ms.
> 
> I have the following setup:
> 
> Asterisk+SendFAX <--SIP/G711--> Asterisk+T38Gateway <---SIP/T38---> ATA
> 
> This T38Gateway application is implemented to send an RTP packet to 
> SendFAX every time it receives an RTP packet.
> 
> On the other side, SendFAX apparently does the same, because as soon as 
> the call is established both Asterisk's exchange RTP packets like crazy 
> - 5000 packets/s.
> 
> What is the best way to solve this problem? How should the voice frames 
> generated in the T38Gateway application to get a correct RTP stream 
> (e.g. 160 samples/frame, 50 frames/s)?

T.38 gateway elements are designed to sit between a packet network and a
TDM network, not between two packet networks. If you are going use
SIP/G.711 as your 'TDM' network, you'll have to force that endpoint to
only generate packets every 20ms, so that it will appear to be a TDM
endpoint. This is not a problem in the gateway application.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming at digium.com
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