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Review request for Asterisk Developers and Joshua Colp.


Bugs: ASTERISK-24563
    https://issues.asterisk.org/jira/browse/ASTERISK-24563


Repository: Asterisk


Description
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When endpoints with direct_media enabled, behind a firewall (Asterisk on a 
separate network) and were bridged sometimes Asterisk would send the ip address 
of the firewall in the sdp to one of the phones in the reinvite resulting in 
one way audio.  When sending the reinvite Asterisk will retrieve the media 
address from the associated rtp instance, but if frames were being read this 
can be overwritten with another address (in this case the firewall's).  This 
patch ensures that Asterisk uses the original device address when using direct 
media.


Diffs
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  branches/12/res/res_pjsip_sdp_rtp.c 428631 
  branches/12/include/asterisk/rtp_engine.h 428631 
  branches/12/include/asterisk/res_pjsip_session.h 428631 
  branches/12/channels/chan_sip.c 428631 
  branches/12/channels/chan_pjsip.c 428631 
  branches/12/bridges/bridge_native_rtp.c 428631 

Diff: https://reviewboard.asterisk.org/r/4216/diff/


Testing
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Used a test bed of 3 phones on a private network behind a firewall with 
Asterisk on another network.  Enabled direct media on the endpoints and then 
had phone A call phone B.  Noted in the logged SIP reinvites that the correct 
address was now being used and also made sure audio flowed in both directions.


Thanks,

Kevin Harwell

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