Hi,

First of all, let me thank you for taking the time to build and update such a complete softphone client.

I have been trying multiple SIP solutions and your seems to be the best I've found. At first, I thought it was an open source solution, but after reading through your website now I clearly understand it's licensed.

And now comes the part when I ask you for your help:

Sadly, I haven't been able to make it work fine with FreeSWITCH under my environment.

I am experiencing a very odd behavior that can be descrived as follows:

REGISTRATION works as Expected. (I am using TLS so capturing the SIP traces becomes complex)

But one of the thing's I've noticed, that looks very odd, is that your REGISTER appears to be sending the wrong port on the "contact:" section. For example, I have 5080 set as the local port, and the contact I am capturing on the PBX looks like this:

Contact: <sip:itazelaar@XXX.XXX.XXX.XXX:5081;transport=tls>;+sip.instance="<urn:uuid:005EFB22-E53E-E511-8ACC-C27007967A46>"

(if I set the port to 5081, then 5082 shows as the contact, etc. looks like there's always 1 added to the original port number)

Anyway, when I try to place a call the PBX gets the INVITE, it forwards the request to the called extension, the phone rings, and I am able to answer. The inbound RTP stream seems to be established fine, but the outbound is never established for some reason.

I am worried about the contact, because I noticed my PBX is trying to send some SIP messages back to that port and, of course, as I am behind a NAT, and firewall, and as your client sets the original port forwarding using UpnP, the port+1 never reaches me.

Let me know if there's anything else you may need to understand this problem properly.

Thanks!
Ivan

My router supports UPnP, it's currently accepting the Phoner Lite commands and it shows all the port mappings. 

My concern comes from the way FreeSWITCH works for TLS (or the way I believe it works, which may be wrong). I've read somewhere on their doc. that their SIP implementation sometimes establishes a secondary TLS channel (PBX to Client) - probably when they need to send something like an INVITE? (I've noticed PhonerLite doesn't ring when I call).
 
Anyway, for these specific situations (PBX-to-Client), the PBX won't validate the client certificate because this is the way I have it configured. Of course, I can set it to validate the client certificate as well but, as you've just said... this has proven to be problematic.

So, first of all, can I confirm you are sending "port + 1" in the contact info? Was this done by design? Is PhonerLite actually listening on that port?

If this is the case... it looks like you are not issuing a UPnP command for the specific "port + 1" (UDP/TCP - depending on the protocol), so any attempts from the outside using the contact info. provided won't make it to PhonerLite... and the original port you have configured will only work for UDP, so in this case, for TLS, it won't work either even if you send the same port on the contact info.

Please let me know if I am wrong or if I have misunderstood something. I know that FreeSWITCH's TLS implementation isn't very standard on this particular aspect (or at least I've understood that), but I guess it should still work... what do you think?

So, going back to PhonerLite not ringing, I guess it's probably caused by FreeSWITCH's implementation trying to open a new TLS TCP stream to the port you are sending. But RTP should still be working, which is odd... 

So as for the RTP, I am not sure why the outbound stream is not being established properly. I don't have a way to log the IP's that are being used other than your own debug window and the SIP traces that I am sending.

Any Ideas on how to trace this? I have tried to configure Wireshark with the keys to decrypt TLS, but I haven't been lucky.

BTW, I am also interested in collaborating with you if you want, first of all to get these specific issues figured out and solved, if needed, through code. I am a skilled dev, with experience in low level programming in assembler (mainly for automation but I've done some interesting things with 8086 embedded on C), C/C++, C#, Java, scripting, among other technologies. 

I would also love to work on the phone UI if you are interested, I am not sure if you are interested, but it will be great to update it's look & feel to a more modern type of UI, but of course, this is only wishfull thinking, I am not sure if I'll have the time, but if you are open to at least give it a try... just let me know.

And hey, thanks for you great support!

No, I am only using TLS for signaling. Haven't set SRTP yet, I am considering it, but haven't done it yet because my PBX's in production now and need to carefully pick the windows to attempt the changes. 

So it is traditional unencrypted RTP for the media streams.

I have made some progress on the wireshark end, for the TLS traces, but it's only bad news, I've confirmed why it's not working  

It looks like FreeSWITCH uses Diffie-Hellman to exchange the secret and therefore, wireshark has no way to decrypt the TLS traffic as it doesn't know the "K" value.

I'll work on having that port forwarding set and let you know how that works, but it will be ideal to have PhonerLite set it using UPnP. Is this something you can consider adding to your solution?

On the RTP, Do you have some recomendations on how to diagnose this? Maybe if I send you the traces on your debug window? I'll dedicate some time to try to diagnose it this monday.

Regarding the collaboration, as you haven't said anything, I'll assume you prefer not to get any more people involved. But if you do, and can use some help, please let me know, it will be great to collaborate on this project!

Thanks!