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You may want to consider fixing RFC 5761 restrictions while you are in there... This prepares for RTP/RTCP muxing in WebRTC "Given these constraints, it is RECOMMENDED to follow the guidelines in the RTP/AVP profile [7] for the choice of RTP payload type values, with the additional restriction that payload type values in the range 64-95 MUST NOT be used. Specifically, dynamic RTP payload types SHOULD be chosen in the range 96-127 where possible. Values below 64 MAY be used if that is insufficient, in which case it is RECOMMENDED that payload type numbers that are not statically assigned by [7] be used first." - Olle E Johansson On Dec. 19, 2014, 9:24 p.m., Scott Griepentrog wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4286/ > ----------------------------------------------------------- > > (Updated Dec. 19, 2014, 9:24 p.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-24367 > https://issues.asterisk.org/jira/browse/ASTERISK-24367 > > > Repository: Asterisk > > > Description > ------- > > Valid payload type codes are between 0 and 127 to allow for being stored in 7 > bits. During call setup, pjsip validates the SDP and will assert if it > encounters an invalid payload type code (see pjmedia_sdp_validate() in > pjmedia/src/pjmedia/sdp.c). This assert will be hit if a call is placed to a > pjsip endpoint with allow=all set. > > To avoid this, the previous use 128 for the slin192 format has been changed > to 95. > > > Diffs > ----- > > /trunk/main/rtp_engine.c 429845 > > Diff: https://reviewboard.asterisk.org/r/4286/diff/ > > > Testing > ------- > > Tested with pjsip calls to allow=all configured extensions. > > > Thanks, > > Scott Griepentrog > >
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