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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