Last week I migrated some of our servers to Asterisk 1.4.18 and we started seeing audio drops of several seconds during SIP calls. After investigating it we noticed that Asterisk was increasing the RTP timestamps abnormally during a conversation. <br>
<br>I'm including a text file with a subset of the data collected by Wireshark that shows the problem (I have the complete packet capture if anybody needs it to analyze it). The Asterisk server is the one whose IP address ends in .38. If you look at the packet with the number 14910 (seq 23369) you'll see that the next packet from Asterisk (14919, seq 23370) increases the RTP timestamp from 77120 to 2280582632. We've tried enabling and disabling internal timing and the jitter buffer, but it made no difference whatsoever. I also added the patch present in ticket #10355 to Asterisk 1.4.18, but it didn't help.<br>
<br>Has anybody else experienced a problem like this one?<br><br><br>