Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

Olle E. Johansson writes: baresip is able to register with two outbound proxies and supports gruu (below). What else is needed? -- Juha # TLS 192.168.43.160:49556 -> 192.168.43.160:5061 REGISTER sip:test.tutpro.com SIP/2.0 Via: SIP/2.0/TLS 127.0.0.1:5047;branch=z9hG4bK305c4adf9d548df2;rport Contact: <sip:test-0x55788569f390@127.0.0.1:5047;transport=tls>;expires=600;+sip.instance="<urn:uuid:9b21c1a8-dc58-a3d6-83d5-7be67cbe70ce>";q=0.5 Max-Forwards: 70 To: <sip:test@test.tutpro.com> From: "Test Account" <sip:test@test.tutpro.com>;tag=f2465ebef6a435a2 Call-ID: ea13feabfb0effd8 CSeq: 1420 REGISTER User-Agent: baresip v1.1.0 (x86_64/linux) Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,REFER Content-Length: 0 # TLS 192.168.43.160:5061 -> 192.168.43.160:49556 SIP/2.0 200 OK Via: SIP/2.0/TLS 127.0.0.1:5047;branch=z9hG4bK305c4adf9d548df2;rport=49556;received=192.168.43.160 To: <sip:test@test.tutpro.com>;tag=75fad08a1e94bbeeabf349d4bd14beda.b29c1116 From: "Test Account" <sip:test@test.tutpro.com>;tag=f2465ebef6a435a2 Call-ID: ea13feabfb0effd8 CSeq: 1420 REGISTER Contact: <sip:test-0x55788569f390@127.0.0.1:5047;transport=tls>;q=0.5;expires=600;received="sip:192.168.43.160:49556;transport=tls";+sip.instance="<urn:uuid:9b21c1a8-dc58-a3d6-83d5-7be67cbe70ce>" Supported: outbound Server: OpenSIPg SIP Proxy (5.6.0-0b02 (x86_64/linux)) Content-Length: # TLS 192.168.43.160:49556 -> 192.168.43.160:5061 INVITE sip:foo@test.tutpro.com SIP/2.0 Via: SIP/2.0/TLS 127.0.0.1:5047;branch=z9hG4bK5b8f364c3b95c0b1;rport Contact: <sip:test-0x55788569f390@127.0.0.1:5047;transport=tls> Max-Forwards: 70 To: <sip:foo@test.tutpro.com> From: "Test Account" <sip:test@test.tutpro.com>;tag=ffb1ab9552ce5a91 Call-ID: 1a70eaa0d9e358bb CSeq: 2779 INVITE User-Agent: baresip v1.1.0 (x86_64/linux) Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,REFER Supported: gruu Content-Type: application/sdp Content-Length: 995