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

2021-06-30 Thread Juha Heinanen
Shahid Hussain writes: > Following are the REGISTER and response messages. Is it possible to > confirm the JSSIP client has full implementation of SIP outbound? Looks like it if you define two sockets. -- Juha __ Kamailio - Users Mailing

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

2021-06-30 Thread Juha Heinanen
Due to other tests, I had missed baresip account's ;outbound paramater. Once I added it, also reg-id was added. -- Juha WSS 192.168.43.160:50442 -> 192.168.43.160:5061 REGISTER sip:test.tutpro.com SIP/2.0 Via: SIP/2.0/WSS 127.0.0.1:9;branch=z9hG4bK5a4ad01f9164d358;rport Contact:

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

2021-06-30 Thread Juha Heinanen
Olle E. Johansson writes: > Full support for SIP outbound (using REG-id when registering etc). > Last time I looked we did not have all nuts and bolts for it, but > let’s give it a try. Yes, reg-id is missing from contact. It would be good to add so that sip proxy can detect if registration is

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

2021-06-30 Thread Juha Heinanen
Olle E. Johansson writes: > > Have you checked baresip? > > I don’t recall baresip having a full SIP outbound implementation. 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

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

2021-06-30 Thread Juha Heinanen
Olle E. Johansson writes: > Full implementation of SIP outbound is the only solution close to > solving this problem in the IETF standards. > However, I have seen no single SIP client that have implemented this, > even though Kamailio supports > it on the server side. The idea is that you

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

2021-06-30 Thread Shahid Hussain
I am using a JSSIP client and they claim to be implemented RFC-5626. Following are the REGISTER and response messages. Is it possible to confirm the JSSIP client has full implementation of SIP outbound? If it supports fully then I can debug outbound and gruu functionality at Kamailio(I have it

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

2021-06-30 Thread Daniel-Constantin Mierla
On 30.06.21 09:28, Juha Heinanen wrote: > Shahid Hussain writes: > >> Would like to know what is the recommended solution for this problem using >> alias or is it a limitation of using alias? > Maybe a limitation. Try with SIP User Agents that support gruu and thus > identify themselves using

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

2021-06-30 Thread Olle E. Johansson
> On 30 Jun 2021, at 10:14, Juha Heinanen wrote: > > Olle E. Johansson writes: > >>> Have you checked baresip? >> >> I don’t recall baresip having a full SIP outbound implementation. > > baresip is able to register with two outbound proxies and supports gruu > (below). What else is needed?

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

2021-06-30 Thread Olle E. Johansson
> On 30 Jun 2021, at 09:49, Juha Heinanen wrote: > > Olle E. Johansson writes: > >> Full implementation of SIP outbound is the only solution close to >> solving this problem in the IETF standards. >> However, I have seen no single SIP client that have implemented this, >> even though

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

2021-06-30 Thread Olle E. Johansson
> On 30 Jun 2021, at 09:10, Shahid Hussain wrote: > > Hi, > Websocket module documentation has a code reference to use aliases for SIP > routing. However, aliases will not work in the following setup and situation. > 1. Kamailio is configured with active and standby node > 2. Ping is