[Asterisk-Users] [patch] Binding rtp to specific interface

[Olle E. Johansson]

http://bugs.digium.com/bug_view_page.php?bug_id=0001019

"This patch allows to bind RTP flows to a specific interface, additionally the SDP session descriptor get's coherent with the same address 
that is used for RTP traffic, this includes sip<->sip and sip<->voicemail and others(not tested, but should work).

Maybe some problems with NAT appear, if anyone notices any bug just tell me :)

The patch applies fine to the CVS version at the time of writing this message."

Please test this patch and report your results to the bug tracker. We need feedback to
judge if this works and if it's ready to be integrated into Asterisk.

Thank you for your input.

/Olle