[asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters

[Michael Maier]

On 21.10.19 at 17:23 Michael Maier wrote:

Attached are actual patches for Asterisk 16.16.0-rc1, 18.0.1 and 18.2.0-rc1 (one 
patch for each version). Version 16.16.0-rc1 is only compile tested (16.14.x was 
the last asterisk 16 version I used myself). 18.2.0-rc1 is used here (as 18.1 and 
18.0.x, too).

All three patches contain a fix for the handling of 183 Session Progress w/o SDP, 
which is handled as 180 Ringing. Another fix handles 180 Ringing w/ SDP as 183 
Session Progress. You may remove them if you don't like them. They are not 
mediasec related. But they reflect "features" of Deutsche Telekom to be sure to 
get ring back and early media for my C610IP.

> How should it all be used now?
> If you want to use SIPS and SRTP with Deutsche Telekom AllIP, you have to be sure to enable the following features in the pjsip trunk (endpoint):
> 
> - transport: tls (TLS 1.2)
> - enable SRTP for this trunk
> - endpoint: support_mediasec=1
> - registration: support_mediasec=1
> 
> 
> 
> If you are using FreePBX, you have to add the support_mediasec switches to
> pjsip.endpoint_custom_post.conf and
> pjsip.registration_custom_post.conf.
> 
> This is done like this:
> 
> File pjsip.endpoint_custom_post.conf:
> [your name of the trunk](+type=endpoint)
> support_mediasec=1
> 
> File pjsip.registration_custom_post.conf:
> [your name of the trunk](+type=registration)
> support_mediasec=true



Thanks
Michael
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