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

[Joshua C. Colp]

On Sun, May 26, 2019, at 1:51 PM, Michael Maier wrote:
> 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?

Encryption of what? If you mean SRTP, there is no association back to the endpoint so you have no idea.

> - 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)?

The ast_sip_session_create_invite function in res_pjsip_session is used to create an INVITE. The SDP generation is in res_pjsip_sdp_rtp.

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
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