[asterisk-users] Skewed RTP timestamps in SIP calls on Asterisk 1.4.18

[Juan Jose Comellas]

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.

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.

Has anybody else experienced a problem like this one?
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