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

[Joshua C. Colp]

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

> - for outgoing REGISTER with authorization header?
> - for outgoing INVITE?

Sessions are handled by res_pjsip_session.c

> 
> To add an additional header, I found the function ast_sip_add_header.
> Would this be the correct function to be used? Can I use this function
> to add more than one header with the same header name?

I haven't tried using it to add more than one header of the same name, but I believe it will work.

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