On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote: > I've got the following setup: > > PhoneA -> > router -> > vpn -> > router-> > asterisk (SIP / ISDN) > > PhoneB -> > asterisk (SIP / ISDN) > > If phone A is connected to phone B (sip-sip), there is a noticable delay > (up to 2-3 seconds) between me speaking and the other end hearing. > > If phone A calls out via the ISDN and back in to the DDI of phone B (ie > SIP->ISDN->ISDN->SIP) then there is no delay at all ! > > Any clues on where I might start looking for this ? >
Are you using canreinvite=yes setting (i.e. is the RTP media expected to flow directly between the phones as opposed to hair-pining through Asterisk)? If so, some of the delay could be attributed to the time spent in RE-INVITEs that happen after the call set up. -- Raj Jain P.S. There is the directrtpsetup= flag that can eliminate this latency (if you're indeed using canreinvite=yes), but I believe that feature is considered "experimental". Actually, if that feature is still experimental, I'd like to change that and fix any associated bugs because it seems like a pretty useful feature to me for people who want to use Asterisk as a call controller (a.k.a. soft-switch) that does not need to participate in the media path. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
