Client to dSipRouter SIP over TLS/STRP <-> PBX with SIP/RTP

[Thomas Kelsey]

I have a dSiprouter set up in front of a FreePBX instance and SIP over UDP and SIP over TLS on the user side to SIP over UDP on the PBX side are working fine if the audio uses

RTP<->RTP proxying to the PBX.  When trying to use SRTP on the client side, dSIProuter appears to just try to negotiate SRTP through to the PBX, which is not currently configured for SRTP.  I'd like to be able to set it up to run SRTP on the client side of dSiprpouter and proxy to RTP on the PBX side.  

Has anyone done this with dSipRouter?  I have done quite a bit of searching and have so far come up empty.  I know it should be possible with kamailio and rtpproxy, but I haven't had time to learn their internals very well yet. 

Thanks,

Thomas Kelsey

[Mack Hendricks]

Hey Thomas,

I think you need to add ;transport=udp on the endpoint within the endpoint group because other wise, Kamailio will just proxy the call through using the same transport as the client.

Sent from my iPhone

On Nov 12, 2020, at 10:54 AM, Thomas Kelsey <tke...@gmail.com> wrote:

I have a dSiprouter set up in front of a FreePBX instance and SIP over UDP and SIP over TLS on the user side to SIP over UDP on the PBX side are working fine if the audio uses

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[Thomas Kelsey]

Thanks, but - I'm pretty sure the ";transport=UDP" just tells it to use UDP for the SIP transport protocol and has no effect on the RTP negotiation.   I'm not having any trouble with SIP/TLS<->SIP/UDP being relayed properly to the PBX.   I'll give it a try to see if it has any effect.

[Mack Hendricks]

Got it…I thought the transport=UDP would trigger the logic in dSIPRouter to tell the RTPEngine to use RTP.  But, maybe we didn’t add that logic.  Let me know.

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[Thomas Kelsey]

I tried it, and it didn't make any difference.  

I did some more detailed traces, and here's what happens regardless of whether the ";transport=UDP" is on the endpoint entry.

The Phone does an initial Invite, goes through the proxy authentication, then sends the final authenticated invite with SRTP in the request.  The invite to the PBX does have the SRTP part of the SDP message stripped out and looks like a normal RTP SDP message.  The PBX sends back a Trying, then OK with a SDP with an agreeing RTP SDP message back.  dsiprouter is then just sending the same SDP payload as received from the PBX through to the phone without SRTP in the OK to the phone.  The phone then ACKs and immediately BYEs the call because it is set up to only accept SRTP.  If I configure the phone for optional or no SRTP, the call completes normally using RTP.

--Thomas Kelsey

[Thomas Kelsey]

It's been a while, and I still have not found a solution to this issue yet.  Are there any recent fixes or improvements in newer builds related to this issue?  I didn't see it in the changelogs but I may have missed it.  I'm on 0.63 currently - have not updated since November.

[Edson Lopes Filho]

Hi there! 

Any news regarding this issue? I am still facing it. It s not possible to get an SIP/TLS EndPoint Registered. Kamailio does not relay Register to PBX in this particular case. UDP/TCP SIP Register works fine. 

[Mack Hendricks]

For Pass-Thru authentication we don’t test this.  So, it probably doesn’t work.  We can definitely put it out to the community and see how many people want this and put it part of our roadmap.  If this is a critical feature for you we can work with you on having it implemented as Professional Services engagement.  Just ping me directly.  

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[Edson Lopes Filho]

Hi, thanks! I will try to figure it out. 

[Mack Hendricks]

Cool….thanks..  Please open a PR when you have it figured out and we will review and accept the PR

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