----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4216/#review13872 -----------------------------------------------------------
branches/12/include/asterisk/rtp_engine.h <https://reviewboard.asterisk.org/r/4216/#comment24364> I'd make this two separate API calls. I'm not a fan of a toggle. I don't think "given remote address" describes what it is enough to be a good name. I think one variable should be renamed to be the "incoming source address" and the other be the "requested target address". - Joshua Colp On Dec. 2, 2014, 11:29 p.m., Kevin Harwell wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4216/ > ----------------------------------------------------------- > > (Updated Dec. 2, 2014, 11:29 p.m.) > > > Review request for Asterisk Developers and Joshua Colp. > > > Bugs: ASTERISK-24563 > https://issues.asterisk.org/jira/browse/ASTERISK-24563 > > > Repository: Asterisk > > > Description > ------- > > 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 > ----- > > 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 > ------- > > 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 > >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
