[asterisk-dev] [Code Review] Pimp SIP Nat

Joshua Colp reviewboard at asterisk.org
Mon Mar 11 06:48:06 CDT 2013


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2363/
-----------------------------------------------------------

(Updated March 11, 2013, 6:48 a.m.)


Review request for Asterisk Developers.


Summary
-------

This change adds a new module, res_sip_nat, which performs rewriting of outgoing messages for cases where address information needs to be replaced when Asterisk is behind NAT. It also performs rewriting of incoming messages for cases where the remote party may be behind NAT. This change also adds ICE support.


Diffs (updated)
-----

  /team/group/pimp_my_sip/res/res_sip_session.c 382783 
  /team/group/pimp_my_sip/res/res_sip_sdp_audio.c 382783 
  /team/group/pimp_my_sip/res/res_sip/sip_distributor.c 382783 
  /team/group/pimp_my_sip/res/res_sip_nat.c PRE-CREATION 
  /team/group/pimp_my_sip/include/asterisk/res_sip.h 382783 
  /team/group/pimp_my_sip/include/asterisk/res_sip_session.h 382783 
  /team/group/pimp_my_sip/res/res_sip/config_transport.c 382783 
  /team/group/pimp_my_sip/res/res_sip/sip_configuration.c 382783 

Diff: https://reviewboard.asterisk.org/r/2363/diff


Testing
-------

Tested Asterisk behind NAT with proper configuration, as well as devices behind NAT. Also tested ICE negotiation.


Thanks,

Joshua

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130311/2976d9ae/attachment.htm>


More information about the asterisk-dev mailing list