----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3439/#review11596 -----------------------------------------------------------
It's great to see this patch, as a larger number of endpoints (particularly of the WebRTC variety) are sending offers with the RTCP attribute. Thanks for adding this! /trunk/channels/chan_sip.c <https://reviewboard.asterisk.org/r/3439/#comment21360> Use sscanf here to process the value, particularly since the value is coming from an external source: if (sscanf(tmp, "%30d", &port) == 1 && port > 0) { /* Process value accordingly */ } - Matt Jordan On April 11, 2014, 8:46 a.m., Olle E Johansson wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3439/ > ----------------------------------------------------------- > > (Updated April 11, 2014, 8:46 a.m.) > > > Review request for Asterisk Developers. > > > Repository: Asterisk > > > Description > ------- > > the A=rtcp attribute in SDP points out a different port than the mediaport+1 > to receive RTCP on. This patch adds a new api to rtpengine and > res_rtp_asterisk and updates chan_sip to use it. > > > Diffs > ----- > > /trunk/res/res_rtp_asterisk.c 412166 > /trunk/main/rtp_engine.c 412166 > /trunk/include/asterisk/rtp_engine.h 412166 > /trunk/channels/chan_sip.c 412166 > > Diff: https://reviewboard.asterisk.org/r/3439/diff/ > > > Testing > ------- > > A massive amount of testing with a test tool for interoperability. > > > Thanks, > > Olle E Johansson > >
-- _____________________________________________________________________ -- 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