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List:       asterisk-dev
Subject:    Re: [asterisk-dev] [Code Review] chan_sip: RFC compliant
From:       "David Vossel" <dvossel () digium ! com>
Date:       2010-06-29 14:30:45
Message-ID: 20100629143045.14425.16097 () hotblack ! digium ! com
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/749/
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(Updated 2010-06-29 09:30:45.485819)


Review request for Asterisk Developers.


Changes
-------

This update reflects comments made by both myself and Nick_Lewis.


Summary
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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.


Diffs (updated)
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  /trunk/channels/chan_sip.c 272920 
  /trunk/channels/sip/include/sip.h 272920 

Diff: https://reviewboard.asterisk.org/r/749/diff


Testing
-------

I tested this with a sipp scenario that sends an INVITE but does not respond to \
Asterisk's 200 OK response.  I verified Asterisk continues to send retransmits until \
the packet times out at the correct timeout time.  I also did a sanity check to \
verify packets continue to be acknowledged correctly by placing some test calls.


Thanks,

David


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