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to another trunk port (same card) that answered the call and transferred it
to another SIP phone. The first analysis shown a good latency time, but
after 1 hour we got about 600ms of latency and after 12 hours we got 11
seconds of latency! It is a unique long duration call. (Asterisk 1.2.22 was
used)

I had done the same test before create the clock source and I thought that
this huge latency was caused by different clocks.

Now, if the clock implemented, I got the same results than before.

Do I misunderstood the problem with SIP/Trunk latency? It is caused by
another problem?

Any help will be appreciated!

Thanks!



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Paulo Garcia

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Hi,<br><br>We created our own clock source using the patched ztdummy approach <font face="Arial" size="2"><span style="font-size: 10pt; font-family: Arial;" lang="EN-US">(<a title="blocked::http://bugs.digium.com/view.php?id=8896" href="http://bugs.digium.com/view.php?id=8896" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
http://bugs.digium.com/view.php?id=8896</a>). It seems to working well, with ztdummy receiving our clock each specified time.<br><br>During our tests, we created the following scenario:<br><br>From SIP Phone we made an outgoing call using a trunk port that was extended to another trunk port (same card) that answered the call and transferred it to another SIP phone. The first analysis shown a good latency time, but after 1 hour we got about 600ms of latency and after 12 hours we got 11 seconds of latency! It is a unique long duration call. (Asterisk 
1.2.22 was used)<br><br>I had done the same test before create the clock source and I thought that this huge latency was caused by different clocks.<br><br>Now, if the clock implemented, I got the same results than before.
<br><br>Do I misunderstood the problem with SIP/Trunk latency? It is caused by another problem?<br><br>Any help will be appreciated!<br><br>Thanks!<br><br><br clear="all"></span></font><br>-- <br>--------------<br>Paulo Garcia
<br><br>

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