[asterisk-users] Asterisk not following SDP port change

[Nick Olsen]

Hello!

I've got a number of asterisk systems running asterisk 16.12.0 currently.
They're configured with PJSIP.
Some of them are behind NAT, some aren't.
All systems have SIP trunks to a Sansay INX.
I've had one-way audio issues calling a particular number. After some
investigation, It seems that the audio is lost due to port changes in SDP.
Other calls don't seem to have any issues. Here's what happens.

Good call:
1. Asterisk sends invite to Sansay INX.
2. Sansay replies 100 Trying without SDP.
3. Sansay replies 183 Progress and provides SDP. This SDP specifies a Media
server and Port. Let's say 192.168.1.2 and port 20100.
4. RTP starts to exchange. Asterisk sending to 192.168.1.2 on port 20100.
5. 200OK is received from Sansay, again it includes SDP showing 192.168.1.2
with port 20100 for RTP.
6. Call works normally.

Bad Call:
1. Asterisk sends invite to Sansay INX.
2. Sansay replies 100 Trying without SDP.
3. Sansay replies 183 Progress and provides SDP. This SDP specifies a Media
server and Port. Let's say 192.168.1.2 and port 20100.
4. RTP starts to exchange. Asterisk sending to 192.168.1.2 on port 20100.
5. 200OK is received from Sansay, This time, It includes modified STP, The
IP is still 192.168.1.2, but the port is now 20180.
6. Asterisk continues to send to SDP 20100.

2nd example of bad call:
1. Asterisk sends invite to Sansay INX.
2. Sansay replies 100 Trying without SDP.
3. Sansay starts sending RTP FROM 192.168.1.2 port 20100
4. Asterisk starts sending RTP to 192.168.1.2 port 20100 (Even with force
rport and comedia, and rewrite contact off, before we ever get the first
SDP).
5. Sansay replies 183 Progress and provides SDP. This SDP specifies a Media
server and Port. Let's say 192.168.1.2 and port 20180 (This is a CHANGE
from step 4).
6. Asterisk is still sending to 192.168.1.2 port.20100 (Ignoring SDP in
step 5 183 progress)
7. 200OK is received from Sansay, Now a third port is received, The IP is
still 192.168.1.2, but the port is now 20250.
8. Asterisk continues to send to SDP 20100. (Ignoring SDP received at step
5 and 7)

I'm hoping I'm missing a simple setting here like "Ignore SDP = no" in
PJSIP. But I've not found it yet.

Secondly, in all cases, Inbound audio (Sansay to asterisk) is fine. And the
source port of RTP DOES change in both bad call examples above. But I
assume asterisk is handling it because the traffic is still arriving on the
same port from asterisk's point of view.

Why calling this one particular number results in 2-3 SDP port changes I
don't yet know. But from what I can see, It's not improper to have port
changes in later SDP. And asterisk *should* follow them. But someone please
correct me if I'm wrong.

Thank you!
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