[Asterisk-Dev] SIP Timer T1 implemented

[Olle E. Johansson]

http://bugs.digium.com/view.php?id=4359

This patch hopefully fixes a lot of timing issues with SIP over bad
networks. There are several reports in the bug tracker related to
timing, and we surely need to continue this work to get timing right
according to the base RFC.

If you have had problems with timing, please test this patch and report
results in the bug tracker.

Technically, we're now retransmitting most messages with more and more
time between the packets until we cancel retransmission. Previous to
this patch, Asterisk retransmits packets with the same pause between
each packet.

If qualify=on for a peer, we are using the measured round-tip time to
schedule re-transmissions (still following the standard).

Thanks for your feedback!

/Olle