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

[Michael Maier]

On 30.05.19 at 10:24 Michael Maier wrote:
[...]
> Another yet missing point is the qualify OPTIONS package. I'm not sure where to add the mediasec headers exactly (which function?). At the
> moment, the Response after OPTION request is (if already registered):
> 
> SIP/2.0 494 Security Agreement Required
> CSeq: 21671 OPTIONS
> Security-Server: msrp-tls;mediasec
> Security-Server: sdes-srtp;mediasec
> Security-Server: dtls-srtp;mediasec
> 
> If you are not already registered, you get a 403 Forbidden.

I managed to add the necessary headers statically in
sip_options_qualify_contact() and they are working as expected. But this 
isn't a good solution. I would like to handle those additions 
dynamically like that:

- sending the usual OPTIONS request
- acting on the 494 Security Agreement Required response
- resend the correct OPTIONS request

Where is the OPTIONS response handled (from a client view) - means, 
where can I check the response code and act on it?


Thanks
Michael