[Asterisk-Users] SIP endpoints ----> RTP stream

Jon Lawrence jon at lawrence.org.uk
Wed Dec 8 12:36:02 MST 2004


On Wednesday 08 December 2004 04:44, Gonzalo Gasca Meza wrote:
> Hi all,
> 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.
>
>
> Example:
> Works like this
> SIP IP phones <-----------Asterisk RTP stream--------------> SIP IP phone
>
>
>                                         Asterisk
>
> SIP IP phones <------------------RTP------------------------> SIP IP phone
>
yes, unless you have canreinvite=no in your sip.conf, assuming that the phones 
negotiate the same codecs then they should be able to initiate a re-invite so 
the the stream goes peer to peer taking * out of the loop.

Jon



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