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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. 2, 2014, 5:29 p.m.)
Review request for Asterisk Developers and Joshua Colp.
Changes
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Addressed findings. This new patch no longer requires changes to the glue
structure and code, but instead stores the explicitly given remote address on
the rtp instance itself. It can then be utilized by the channel drivers where
appropriate. Currently, the only time the given_remote_address and original
remote_address should differ is when the rtp engine "learns" a new address and
symmetric rtp is enabled.
Re-tested everything with the new patch including chan_sip this time.
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 428786
branches/12/main/rtp_engine.c 428786
branches/12/include/asterisk/rtp_engine.h 428786
branches/12/channels/chan_sip.c 428786
branches/12/channels/chan_pjsip.c 428786
branches/12/addons/chan_ooh323.c 428786
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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