[asterisk-users] does res_pjsip support ZRTP?

Matthew Jordan mjordan at digium.com
Tue Oct 6 08:08:00 CDT 2015

On Mon, Oct 5, 2015 at 3:58 PM, Dmitriy Serov <serov.d.p at gmail.com> 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.
> - 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.

I know this is shocking to hear, but this is an open source project.

That means anyone can fix something. Anyone can add something. Even
you! You have all the power to affect your system.

It also means that no one is under any obligation to do it for you.

Surprising, right? I know, it's amazing to think that *YOU* have all
the responsibility and power.

We use PJSIP. We use it in a variety of settings. It works well for
us. Does that mean it works well for you? I don't know. I'm not you. I
don't have your use cases. Would I like it to work well for you? Of
course! But if you don't participate by reporting issues, testing
changes, and contributing code, there's not much I can do for you
other than to note that the line is long, and feel free to stand in it
until someone in the community gets around to what you'd like to have


Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

More information about the asterisk-users mailing list