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
Thanks!
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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 This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041208/b3f5da39/attachment.pgp
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@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
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