Problem with websocket and sip messages


Firstly Merry Christmas to all.

I have 2 standard udp SIP phone (A, B) with 2 webRTC SIP Phone (C and D).
When A and C are in communication there is not problem to send SIP (several SIP message) message from A to C. But where B and D are in communication to, many sip message are not send from Asterisk to D or from Asterisk to C.
The Sip messages arrive normaly from A and B to Asterisk but many are drop from Asterisk to C or to D.

Could you have any idea to understand and / or to solve this problem.


Asterisk is a back to back user agent, not a proxy.

That means it terminates a SIP session on the incoming side and runs a completely different session, which might not even be SIP, on the outgoing side. No details of the SIP request contents are passed through the core of Asterisk.

There are certain circumstances in which it can take advantage of both sides being SIP and bypass the core, but those mainly relate to media handling.


I understand, but why all SIP messages are correctly send from A (udp SIP) to Asterisk and from Asterisk to C(websocket SIP) when they are alone. But when I add B(udp SIP) and D(websocket SIP) many SIP messages are drop between A and C and also between B and D.


I’m assuming by message you mean request or response, not MESSAGE requests.

For the former, you should not be relying on requests and responses flowing end to end.

For the latter, you haven’t provided enough information for me to understand your configuration and exactly what you are doing.