[Linphone-users] Why is Linphone apparently mixing SIP and SIPS schemes?

[Javier Fernández Aparicio]

Hi everyone, 

I'm building a PABX system with Asterisk and FreePBX and I need trustworthy phones to perform tests, that's why I'm using Linphone. The problem is that I need to secure all communications so I'm using `sips` for the URIs, however, I can see in the asterisk logger a mix between sip and sips schemes.

Why is this happening? Apparently once I use the sips scheme all communications should be forced to also use sips.

As you can see in this asterisk log https://jfernandz.me/~wyre/linphone-linphone_2.log, both contacts (phones) are apparently registered using the sips scheme:

asc3*CLI> pjsip show contacts

  Contact:  <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================

  Contact:  035000/sips:035000@10.100.0.26:41856;transport b131d6fe7e Avail       105.646
  Contact:  052002/sips:052002@10.100.0.4:45504;transport= 2a6084e8d3 Avail       131.221

Objects found: 2

I've placed a test call between two linphone clients and you can see a mix between sip and sips schemes, for example, I can see

Contact: <sip:035000@10.100.0.26:41856;transport=tls>;expires=599;+sip.instance="<urn:uuid:5e85b13b-b9e6-008a-b342-fdcb81341b3a>";+org.linphone.specs="ephemeral,groupchat/1.1,lime"

I think this could be causing the placed call is hungup immediately. Is there some way to force sips usage in your Linphone client app? 

Thank you all. BR.

Javier.

Javier Fernández Aparicio DevOps Engineer Headquarters - Paseo Castellana, 200 - SPACES - Madrid 28046 ES Labs - Calle Innovación, 17 - Getafe, Madrid 28906 ES https://www.joifilabs.com/