David Vossel 23b6e621d2 chan_sip: RFC compliant retransmission timeout ... Retransmission of packets should not be based on how many packets were sent, but instead on a timeout period. Depending on whether or not the packet is for a INVITE or NON-INVITE transaction, the number of packets sent during the retransmission timeout period will be different, so timing out based on the number of packets sent is not accurate. This patch fixes this by removing the retransmit limit and only stopping retransmission after a timeout period is reached. By default this timeout period is 64*(Timer T1) for both INVITE and non-INVITE transactions. For more information on sip timer values refer to RFC3261 Appendix A. Review: https://reviewboard.asterisk.org/r/749/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276219 65c4cc65-6c06-0410-ace0-fbb531ad65f3		2010-07-13 22:18:38 +00:00
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include	chan_sip: RFC compliant retransmission timeout	2010-07-13 22:18:38 +00:00
config_parser.c	Kill some startup warnings and errors and make some messages more helpful in tracking down the source.	2010-07-09 17:00:22 +00:00
dialplan_functions.c	Kill some startup warnings and errors and make some messages more helpful in tracking down the source.	2010-07-09 17:00:22 +00:00
reqresp_parser.c	Kill some startup warnings and errors and make some messages more helpful in tracking down the source.	2010-07-09 17:00:22 +00:00
sdp_crypto.c	Add SRTP support for Asterisk	2010-06-08 05:29:08 +00:00
srtp.c	Add SRTP support for Asterisk	2010-06-08 05:29:08 +00:00