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

[Joshua C. Colp]

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:
> > 
> > <snip>
> > 
> >>
> >>
> >> If I wanted to try it myself - what would be the correct places to
> >> implement it?
> >>
> >> It shouldn't be that complicated, because it seems mostly to be done by
> >> adding some additional headers during different states and check for
> >> them in the answers. The rest should be mostly the same as used for
> >> existing SRTP.
> >>
> >> 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()?
> 
> >> - for outgoing REGISTER with authorization header?
> 
> handle_registration_response()?
> 
> >> - for outgoing INVITE?
> > 
> > Sessions are handled by res_pjsip_session.c
> 
> new_invite()?
> 
> I additionally need to add two dsp entries:
> a=3ge2ae:requested                     |
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:EpcgtOdT5qd...
> 
> Which function would be the best to add these sdps?

The functions you provided would be the correct places at a glance. For SDP it's handled in the res_pjsip_sdp_rtp module, following where crypto is added would be the easiest option for there.

-- 
Joshua C. Colp
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