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

[Iqbal Gandham]

Yup. Asterisk was built to handle the media stream, what you are looking 
for is aproxy, which is what SER is best at doing. But then you may have 
a problem in billing.

Iqbal

Tracy R Reed wrote:
> 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
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users