On Wed, Mar 3, 2021 at 5:55 PM Nick Olsen <n...@141networks.com> wrote:
> > SDP for the first 183 > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): Sansay-VSXi 188 1 IN IP4 > XX.XX.XX.12 > Session Name (s): Session Controller > Connection Information (c): IN IP4 XX.XX.XX.46 > Time Description, active time (t): 0 0 > Media Description, name and address (m): audio 14996 RTP/AVP 0 > 101 > Media Attribute (a): rtpmap:0 PCMU/8000 > Media Attribute (a): rtpmap:101 telephone-event/8000 > Media Attribute (a): fmtp:101 0-15 > Media Attribute (a): ptime:20 > Media Attribute (a): sendrecv > > > SDP for the 2nd 183 > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): Sansay-VSXi 188 1 IN IP4 > XX.XX.XX.12 > Session Name (s): Session Controller > Connection Information (c): IN IP4 XX.XX.XX.46 > Time Description, active time (t): 0 0 > Media Description, name and address (m): audio 15104 RTP/AVP 0 > 101 > Media Attribute (a): rtpmap:0 PCMU/8000 > Media Attribute (a): rtpmap:101 telephone-event/8000 > Media Attribute (a): fmtp:101 0-15 > Media Attribute (a): ptime:20 > Media Attribute (a): sendrecv > > SDP for the 200OK. > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): Sansay-VSXi 188 1 IN IP4 > XX.XX.XX.12 > Session Name (s): Session Controller > Connection Information (c): IN IP4 XX.XX.XX.46 > Time Description, active time (t): 0 0 > Media Description, name and address (m): audio 15252 RTP/AVP 0 > 101 > Media Attribute (a): rtpmap:0 PCMU/8000 > Media Attribute (a): rtpmap:101 telephone-event/8000 > Media Attribute (a): fmtp:101 0-15 > Media Attribute (a): sendrecv > Media Attribute (a): ptime:20 > > Still working on the logs, But gather anything from that so far? > > In this case, asterisk always sent to the first provided RTP port of 14996. > One thing that does stand out is they aren't obeying the RFC, as the version number in the o line should be incremented[1]. PJSIP is more tolerant of that though I believe. It did jog my memory though on an option[2][3] which may apply here. You'll want to set it both in system and on the endpoint. [1] https://tools.ietf.org/html/rfc4566#page-11 [2] https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L1096 [3] https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L889 -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org
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