SIP Proxy example didn't implement INVITE about sipsorcery HOT 4 OPEN

You're trying to build something really complicated, and I think trying to use this library to do something like that is unlikely to lead to success.
As I understand it you want to answer a call, do some IVR, and the forward the call to to something.
There are products like asterisk/freeswitch and kamailio that exist to do stuff like this. Those are not simple solution either. But if you want to comply with standards it's probably worth looking into it.

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No, I don't want to do some IVR and then forward the call. I just want to forward the call immediately, in other words just act exactly like a Proxy Server by using the Contact headers. This is not a complex use case, I feel like it's 15 lines of code on top of the massive standards-compliant library you guys have already built.

The next server that I forward it to is the one that has the IVR. The IVR destination server is also based on this library, it would be great if I could make the Proxy Server based on this library too, then I would have less variety of products to support and it would all be in .Net.

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Thanks for the explanation, that makes more sense now. Just to clarify, I'm a minor contributor to this project, and by no means a sip expert like the maintainers.

I'm not sure if sipsorcery respects record-route headers when sending an ACK, or if it replies to the contact in the sip invite ok. And indeed I'm not actually sure what it's supposed to do according to the standard. I recently had a similar problem in this area and couldn't bottom it out so implemented a workaround for my scenario.

To try to understand what it's doing I would use wireshark on the client and see where it's trying to send the ack to. That might give you some clues.

Then if you can include the full sip messages for your flow somebody more knowledgeable might be able to help you out with what eitherinyou are doing wrong, or what might need implementing in sipsorcery.

I'll be watching this in the hope somebody else jumps in with an answer. Routing in sip is hard to get your head around.

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Sounds like you're attempting to build a Stateful Proxy (like OpenSER, Kamailio) or a B2BUA Server (like Asterisk and FreeSWITCH w/o media).

Both options are feasible with this library but definitely not in 15 lines of code, probably closer to 1500.

There is an example of a B2BUA here.

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