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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4216/
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(Updated Dec. 4, 2014, 3:31 p.m.)
Review request for Asterisk Developers and Joshua Colp.
Changes
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Changed to rename things only be internal to the rtp_engine source. Any
externally needed api calls were added instead of changed.
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 (updated)
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branches/12/res/res_rtp_asterisk.c 428862
branches/12/main/rtp_engine.c 428862
branches/12/include/asterisk/rtp_engine.h 428862
branches/12/channels/chan_sip.c 428862
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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