[sr-dev] SIP to websocket routing problems

Hello everybody, we've a kamailio setup as a SIP proxy/SBC that we use to route incoming request into our system . We also need to support web browser clients, so, we properly enabled and configured the websocket module according to its module's documentation: https://www.kamailio.org/docs/modules/devel/modules/websocket.html#idm37 This solution is working fine for both TLS SIP clients and incoming websocket INVITEs but, when we try to route a SIP call from a SIP client to a registered websocket client, our SBC is unable to find the proper socket connection even if the proper IP, port and transport is received in the Request-URI or the Route header: Received request URI looks like: * INVITE sip:b80snduq@185.41.96.32:54510;transport=wss SIP/2.0 while the Route header looks like: * Route: <sip:10.122.144.132:5061;transport=tls;lr;received=sip:185.41.96.32:54510%3Btransport%3Dws> My only guess is that the transport parameter, for some reason, is not being properly parsed but I've been debugging core/forward.c with no luck as it seems that the request is not even being forwarded. Any suggestion about what code I should look into? Is the destination IP, port and protocol usually set in the core parsing or might this be a nathelper module error? Thank you very much in advance for your help. BR, Joel.