On 15-10-05 05:58 PM, Dmitriy Serov wrote:
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.
Can you clarify what you mean here? There's an anonymous endpoint
identifier which can be used for anonymous inbound messages basically.
- found hardphones and software phones that don't accept "long nonce"
and refuse to register when using res_pjsip
Have you filed an issue with this and details about the
hardphones+softphones?
- enable icesupport also leads to problems of registration and cannot be
"common solution"
icesupport is only applied to calls, what happens for registration?
- 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.
Have you filed any issues for these with information? We can't make
PJSIP better if we don't know about the problems people are having like
this.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
