[asterisk-dev] [Code Review] 4032: PJSIP: Force transport on contact rewrite

opticron reviewboard at asterisk.org
Mon Sep 29 18:53:41 CDT 2014


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4032/
-----------------------------------------------------------

(Updated Sept. 29, 2014, 6:53 p.m.)


Review request for Asterisk Developers.


Changes
-------

Address Josh's comment.


Repository: Asterisk


Description
-------

If contact rewriting is enabled but the contact differs in transport from what is actually being used, messages after the initial INVITE transaction can be sent to an incorrect transport/port combination. In the case where this bug occurred the remote party never received a BYE since it was sent to the remote party's TCP port over UDP.


Diffs (updated)
-----

  branches/12/res/res_pjsip_nat.c 424094 

Diff: https://reviewboard.asterisk.org/r/4032/diff/


Testing
-------

Ensured that this patch allowed the BYE to be sent properly.


Thanks,

opticron

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140929/ba6a6f26/attachment.html>


More information about the asterisk-dev mailing list