[Asterisk-Users] SIP endpoints ----> RTP stream
Tracy R Reed
treed at copilotconsulting.com
Wed Dec 8 00:21:44 MST 2004
On Tue, Dec 07, 2004 at 08:44:50PM -0800, Gonzalo Gasca Meza spake thusly:
> I have just setup Asterisk, but the problem is that all RTP stream pass through Asterisk, I mean all call setup and voice stream pass trough Asterisk once the call is established.
> Is there a way that call setup is established, the RTP stream pass just between the SIP endpoints.
Yes. For this you should use SER:
www.iptel.org/ser
--
Tracy Reed http://copilotcom.com
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