[asterisk-users] strange SIP-SIP delay
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 ? Julian ___ -- 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
Re: [asterisk-users] strange SIP-SIP delay
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 ? > > Julian > Have you tested the latency across your VPN? Thanks, Steve T ___ -- 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
Re: [asterisk-users] strange SIP-SIP delay
Hi Steve - the vpn is a "consistent" as the sip->IDSN has to go through the VPN first to get to asterisk. i.e. to make an "outside" call, PhoneA goes through the vpn to the asterisk box, and out through isdn. Julian Steve Totaro wrote: > 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 ? >> >> Julian >> > > Have you tested the latency across your VPN? > > Thanks, > Steve T > ___ -- 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
Re: [asterisk-users] strange SIP-SIP delay
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
