SIP call with SRTP doesn't work, RTP fallback doesn't work either

Describe your environment

• Device model: Samsung SM-T720
• Android version: Android 11 API30
• What build you are using: your own android: 94f816d3 daemon:3c25f607f4f52028409fc0ea119ca943550e55e0

Steps to reproduce

• Can you reproduce the bug: at will
• Steps:
  1. Active param Encrypt media streams (SRTP) 2. Try to place or receive a call
• Actual result: it doesn't work 3. Active param Fallback to RTP if SRTP fails
• Actual result: it doesn't fallback

added BacklogMedium SIP bug labels

marked this issue as related to #1312 (closed)

What's your configuration server side?

In VoIP.ms, Account settings, Encrypted SIP traffic (Yes/No):

• If enabled, in Jami, you need to enable TLS + SDES key exchange (if no SDES, the call will fail)

So SDES is important and taken into account

Also on our Asterisk I'm pretty sure TLS and SDES are not configured (only UDP & RTP)

However for fallback, seems unchecked since ~8 years. So even before I work on Jami

Pierre have you check the port configuration if you have change it for your sever side, I have notice every time you restart the app the ports get back to the default configuration in the SIP account

mentioned in issue #1340

changed the description