[SR-Users] Kamailio + RTPEngine SIP Proxy...

[Richard Edmands]

I know this works with an Asterisk configuration, not sure how
Freeswitch is any different yet. Haven't needed to poke Freeswitch.

https://github.com/fredposner/scripts/blob/master/kamailio/kamailio-tls_srtp-bridge.cfg
Strip out the fun bits where Fred is re-writing the contact header.

Then in the route relay, add the following:
record_route();
handle_ruri_alias();
add_contact_alias();

In some configs, onreply_route manage_reply needs a:
add_contact_alias();

Lazy, yes. Works for me?, yes.

On Mon, 5 Sept 2022 at 14:16, Jerry Kendall
<Jerry.Kendall at bishophosting.com> wrote:
>
> Hi There,
>
>
> It's been a few years since I joind the group and I have watched and learnt and played but, I am stuck...
>
> I have searched hi & low but I can't seem to find what I am looking for.
>
>
> Is is do-able to have SIP Phones proxy through Kamailio/RTPengine and have the connection relayed to a freeswitch system where all the work is done.
>
> I need to have Kamailio act as proxy ONLY - no need to do AUTH, and I need Kamilio to have the RTPEngine have the media but only as proxy.
>
> I have FreeSWITCH doing all the IVR/VM/CONF/etc....
>
>
> So:
>
> 1) The SIP Phone sends everything to the Kamailio system, the Kamailio system will relay everything to the FreeSWITCH system (requests and replies)
>
> 2) The FreeSWITCH system will send everything to the Kamailio system which will relay everything back to the SIP Phone (requests and replies)
>
> 3) The SIP phones are behind NAT
>
> 4) The Kamailio and FreeSWITCH systems are on the public internet with public IP addresses
>
>
> I have found a few sample kamailio.cfg files but, I can get REG to make it through but no SUBCRIBE wont get NOTIFYs back, etc...
>
> I seem to get parts but not all the way.
>
>
> If this is doable, and I assume it is, is there a config file someone can share with me? or point me to?
>
>
> Jerry
>
>
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