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

Michael Maier m1278468 at mailbox.org
Fri May 24 02:28:57 CDT 2019

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

>> - for outgoing REGISTER with authorization header?


>> - for outgoing INVITE?
> Sessions are handled by res_pjsip_session.c


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?


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