[asterisk-users] Asterisk not following SDP port change

[Joshua C. Colp]

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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