Pavel Jezek wrote: > > Steve Murphy wrote: >> Hello-- >> >> Why do I target chan_sip for so much effort? Because, >> it seems to me, chan_sip is probably the most used channel >> driver in the asterisk community!! (and, of course, >> the zap/dahdi driver, is also pretty popular) >> >> I haven't had time to follow up on chan_sip, and I probably >> won't for several months. >> >> But, if I had time, here is what I'd do: >> >> There are two ways to speed up chan_sip, and they are separate issues, >> tied together on how many cpu cycles they use up: >> >> 1. Call setup/teardown (invites/hangups) -- limits the calls/sec >> asterisk can handle. >> > > one of the big issues in sip callsetup performance, that appears to me > in current trunk, is about 500ms delay in propagation SIP/OK message > between bridged parties > eg.: one party answers call, send SIP/OK with SDP to asterisk, asterisk > then forwards it to other party, but with unacceptable delay about 500ms! > this is so much, that users complaining about lost first word of speech > communication, > I posted info about this to bugreport, that seems to be related to this, > look at my message: > http://bugs.digium.com/view.php?id=12708#91173 > I also attach graph picture from wireshark, that clearly ilustrated, > where is problem.... (OK-SDP-delay.png ) > PJ >
If you check ast_answer in channel.c of trunk, you can see that it calls __ast_answer(chan, 500). The 500 there is a 500 ms delay that occurs before calling the channel's answer callback. In the case of SIP, this would indeed mean that there is a 500 ms delay between receiving the 200 OK from the callee and sending a 200 OK to the caller. Mark Michelson _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
