Hello everyone, I’m having some problems routing a MESSAGE from Asterisk to a WebRTC client. Regular calls work fine, but MESSAGE handling is failing. Kamailio: 10.5.0.8 Asterisk: 10.5.0.2
I suspect the problem could be one of two things: ->The CSeq values don’t match (CSeq: 2 MESSAGE vs CSeq: 49306 MESSAGE). ->The To and From headers might be preventing Kamailio from knowing where to send the reply. Has anyone seen this behavior before? Any suggestions on how to correctly route MESSAGEs through Kamailio to WebRTC clients? Here are some logs: MESSAGE sip:[email protected] SIP/2.0 Transport: udp Record-Route: <sip:172.31.217.74:5001;r2=on;lr;ftag=124pqf5crm;nat=yes> Record-Route: <sip:172.31.217.74:8087;transport=ws;r2=on;lr;ftag=124pqf5crm;nat=yes> Via: SIP/2.0/UDP 172.31.217.74:5001;branch=z9hG4bK43b3.e83a57badbec3df404fc69a3c7f0e666.0 Via: SIP/2.0/WSS ip9sqfgevvko.invalid;rport=59806;received=172.31.208.1;branch=z9hG4bK324262 To: <sip:[email protected]> From: "916c67e8-652f-49ea-b288-2bbbee8dbf21" <sip:[email protected]>;tag=124pqf5crm CSeq: 2 MESSAGE Call-ID: s1djti0r9egfcvi1n4aa Max-Forwards: 69 Authorization: Digest algorithm=MD5, username="User1", realm="xxxxx", nonce="1756131542/ed3931083af68713eb56098000b661ec", uri="sip:[email protected]", response="3a474fc4f735e602a7b1940d0c9da564", opaque="10b0b5293ff53cc5", qop=auth, cnonce="0reui7po4gfn", nc=00000001 Supported: outbound User-Agent: SIP.js/0.21.1 Content-Type: text/plain Content-Length: 68 {"message":"ola","sessionId":"feff3811-f3b4-4dd7-8125-af9fe80dd523"} Some log from asterisk: pjsip:User1/sip:[email protected]:5001;x-ast;transport=udp,"MaxTalk" <sip:[email protected]>,"ChatApp xxxxx" <sip:[email protected]:5001;x-ast>" <--- Transmitting SIP request (687 bytes) to UDP:10.5.0.8:5001 ---> MESSAGE sip:[email protected]:5001;transport=udp;x-ast SIP/2.0 Via: SIP/2.0/UDP 10.5.0.2:4040;rport;branch=z9hG4bKPj782df7e5-c4b7-4f74-98ff-611d39069cc5 From: "MaxTalk" <sip:[email protected]>;tag=645539f3-3177-436a-88f6-ac7f64735fd4 To: "ChatApp xxxxx" <sip:[email protected];x-ast> Contact: <sip:[email protected]:4040> Call-ID: 2af343eb-67f5-4458-ba07-37b830169ec7 CSeq: 49306 MESSAGE Max-Forwards: 70 User-Agent: Asterisk PBX 18.24.3 Content-Type: text/plain Content-Length: 138 Olá! Como posso ajudar-te hoje? Se tiveres alguma dúvida sobre a plataforma da xxxxx ou qualquer outra questão, estou aqui para ajudar. <--- Received SIP response (476 bytes) from UDP:10.5.0.8:5001 ---> SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 10.5.0.2:4040;received=10.5.0.2;rport=4040;branch=z9hG4bKPj782df7e5-c4b7-4f74-98ff-611d39069cc5 From: "MaxTalk" <sip:[email protected]>;tag=645539f3-3177-436a-88f6-ac7f64735fd4 To: "ChatApp xxxxx" <sip:[email protected];x-ast>;tag=19162a7003f0e5d10acb0ff84f0e52ca.11182266 Call-ID: 2af343eb-67f5-4458-ba07-37b830169ec7 CSeq: 49306 MESSAGE Server: kamailio (5.7.6 (x86_64/linux)) Content-Length: 0 Thank you. __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- [email protected] To unsubscribe send an email to [email protected] Important: keep the mailing list in the recipients, do not reply only to the sender!
