[asterisk-dev] Regarding SIP performance

[Vlasis Hatzistavrou]

> At the moment (SVN), Asterisk is passing on an INVITE with SDP pointing to
> the first endpoint. That works sometimes, but not other times, depending > > on which codecs the second endpoint supports. But you have to know whether > to try this before you send on the INVITE.


Perhaps this could be solved by adding a new option in the Dial command? This will allow for other channel drivers or protocols to use the option, because direct RTP connections are also useful for H323 and MGCP...

Best regards,
Vlasis Hatzistavrou.