[asterisk-users] does res_pjsip support ZRTP?

Dmitriy Serov serov.d.p at gmail.com
Mon Oct 5 15:58:54 CDT 2015

05.10.2015 23:24, Joshua Colp пишет:
> On 15-10-05 05:22 PM, Dmitriy Serov wrote:
>> Hello. Do I understand correctly that the current implementation
>> res_pjsip does not support ZRTP?
>> http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html
> ZRTP is not supported in Asterisk itself.
>> Nothing has changed since 2013? P.S. I greatly regret that moved from
>> chan_sip to res_pjsip. Previously used very much lacking, and much of
>> the promise failed. Dmitriy Serov.
> Any particular examples?

- opus support. Ok... I know the reason why it is not supported fully 
this codec. But the existing foreign solution works fine with chan_sip 
and does not work with res_pjsip works.
- endpoint specific ACL
- No support for SIP message without authorization. For this reason, the 
previously working functionality of sending and receiving SMS from 
gateway GOIP had to rewrite their internal Protocol.
- found hardphones and software phones that don't accept "long nonce" 
and refuse to register when using res_pjsip
- enable icesupport also leads to problems of registration and cannot be 
"common solution"
- issue tracker now contains multiple error messages that arise every 
day and reboot my server (which cannot be called a production)
- And watchdog logs SegFaults and Hangs including other stacks that are 
not yet documented in the issue tracker.

Be sure to have forgotten something, because it is not documented all 
meet and unsolved problems,workarounds.

The transition to PJSIP was chosen as mainstream and full support for 
WebRTC. As a result, instead of developing a service I a few months I'm 
returning opportunities to which users are accustomed and expect to see.
Having the knowledge and the overall picture a few months ago I would 
not have taken such a decision.

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