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

[Michael Maier]

On 24.05.19 at 12:45 Joshua C. Colp wrote:
> On Fri, May 24, 2019, at 3:35 AM, Michael Maier wrote:
>> On 03.02.19 at 12:00 Joshua C. Colp wrote:
>>> On Sun, Feb 3, 2019, at 6:03 AM, Michael Maier wrote:
>>>> On 15.01.19 at 20:27 Joshua C. Colp wrote:

[...]
>>>> In which function should the headers been added?
>>>> - for outgoing initial REGISTER?
>>>
>>> Outbound registration is handled by res_pjsip_outbound_registration.c
>>
>> Could you please suggest the function name, which should be started at 
>> best? Would it be
>> handle_client_registration()?

- Is there any possibility in handle_client_registration to check if encryption is enabled?
- How do I know if the registration performed is the initial registration or a subsequent
  registration after EXPIRE?

>>>> - for outgoing REGISTER with authorization header?
>>
>> handle_registration_response()?
>>
>>>> - for outgoing INVITE?
>>>
>>> Sessions are handled by res_pjsip_session.c
>>
>> new_invite()?

This function seems not to be called when a new outgoing INVITE is started. Do you have any idea which function could be used (asterisk 16)?


Thanks
Michael