[asterisk-users] Asterisk not following SDP port change

[Nick Olsen]

accept_multiple_sdp_answers=yes fixed it.

It now follows SDP a total of 3 times in my tests.

I had found this setting before posting. And had toggled it. But it didn't
make any difference until defined in system (in addition to the endpoint
itself).

Thanks for your help!

On Wed, Mar 3, 2021 at 5:21 PM Joshua C. Colp <jcolp at digium.com> wrote:

> On Wed, Mar 3, 2021 at 5:55 PM Nick Olsen <nick at 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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