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

Olle E. Johansson oej at edvina.net
Wed Jun 30 10:02:34 CEST 2021

> On 30 Jun 2021, at 09:49, Juha Heinanen <jh at tutpro.com> 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 Kamailio supports 
>> it on the server side. The idea is that you always have two TCP
>> connections to two active servers.
> Have you checked baresip?

I don’t recall baresip having a full SIP outbound implementation.

SIP outbound seems to be an elegant solution to a non-problem. Maybe it’s coming back on the radar now, several years later and we need to start working on it as more and more SIP moves to TCP/TLS or WebSocket and network operators implement CGnat and other middleboxes that do strange stuff to open flows.

Without SIP outbound a SIP UA using TLS needs to have a server cert and accept incoming TLS connections, which of course does not work over NAT, so if we want to be RFC compliant, we should at least copy the RFC for SIP over WebSocket “half-outbound” requirement to SIP/TLS and SIP/TCP.

I have started discussions about this “half-outbound” concept a few times, but haven’t gotten much support in the SIPcore IETF group. Meanwhile, Kamailio actually breaks the RFC by sending outbound requests on the inbound TLS connection. It works, but it’s not RFC compliant.


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