I have ethereal installed and I'll do a full call trace. The Catch 22 is I don't have access to access to a source of repeatable (ie recorded) content accessed through IAX. That would help in producing traces for the ATA and 7960 for comparison. I mainly use IAX for non-critical international business calls to people who wouldn't want to be * testers.
Iain
--On Tuesday, May 18, 2004 7:22 pm -0600 "brian k. west" <[EMAIL PROTECTED]> wrote:
Lets look at this and FIX the problem instead of hacking it. What you need to do is install etherreal and capture a call and parse the timestamp info to see if they are slipping. Because they are perfect here.
bkw
----- Original Message ----- From: "Brian Cuthie" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, May 18, 2004 5:07 PM Subject: Re: [Asterisk-Users] AArgh, * and the 7960
OK?
Iain,
This is a known issue with the Cisco phone and Asterisk having to do with a change made later in the cvs tree. Try 1.0 stable, or modify rtp.c to comment out the two lines as follows:
/* Re-calculate last TS */ rtp->lastts = rtp->lastts + ms * 8; // if (!f->delivery.tv_sec && !f->delivery.tv_usec) { /* If this isn't an absolute delivery time, Check if it is close to our prediction, and if so, go with our prediction */ if (abs(rtp->lastts - pred) < 640) rtp->lastts = pred; else { ast_log(LOG_DEBUG, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms); mark = 1; } // } } else {
This seems to work for me. Others may have more insight.
-brian
Nik Martin wrote:
> Out of context, this isn't much information. Is your network > connection> Is your broadband provider having troubles? Has some upstream hardware > changed that you may not be aware of? > > > > >> -----Original Message----- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of >> Iain Stevenson >> Sent: Tuesday, May 18, 2004 1:29 PM >> To: [EMAIL PROTECTED] >> Subject: [Asterisk-Users] AArgh, * and the 7960 >> >> >> >> I've just had the most appalling performance from * ever. Dialling: >> >> Cisco 7960 => asterisk => IAX >> >> produces sound drop outs so extreme that the call is useless. >> I noted this >> in an earlier post. Dialling: >> >> Cisco ATA186 => asterisk => IAX >> >> is fine. >> >> Frankly, I think this is such a bad problem that it should be >> sorted in >> advance of any of the new features that seem to be getting >> such prominence >> nowadays. It was not present earlier in the year and I >> haven't upgraded my >> 7960. So I don't think you can point the finger entirely in Cisco's >> direction. >> >> Iain >> _______________________________________________ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
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