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Does chan_sip also have this problem? I'd assume so (just wondering) branches/12/channels/chan_pjsip.c <https://reviewboard.asterisk.org/r/4216/#comment24335> This is an incorrect assumption and won't work. Either side may be RTP capable - meaning there is no guarantee that both will be PJSIP. branches/12/channels/chan_pjsip.c <https://reviewboard.asterisk.org/r/4216/#comment24336> I also don't like that you are overloading the purpose of this callback. It's meant to determine if the two sides will allow a remote RTP or not. I think this is too focused on chan_pjsip. Realistically this problem can happen with any channel driver that uses the RTP engine and res_rtp_asterisk. Something generic should be done instead, imo. - Joshua Colp On Nov. 26, 2014, 10:17 p.m., Kevin Harwell wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4216/ > ----------------------------------------------------------- > > (Updated Nov. 26, 2014, 10:17 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_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 > ------- > > 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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