9 thoughts on - Does Res_pjsip Support ZRTP?

• ZRTP is not supported in Asterisk itself.

  Any particular examples?

• 05.10.2015 23:24, Joshua Colp пишет:
  – 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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• Can you clarify what you mean here? There’s an anonymous endpoint identifier which can be used for anonymous inbound messages basically.

  Have you filed an issue with this and details about the hardphones+softphones?

  icesupport is only applied to calls, what happens for registration?

  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: http://www.digium.com & http://www.asterisk.org

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• 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 done.

  Matt

  —
  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

  —

• 06.10.2015 16:08, Matthew Jordan пишет:
  I these words were repeatedly read and remember them well. That’s why I
  haven’t written any complaints to the developers. Where you saw them, what made again to write these words?
  I am a developer with more than two dozen years of experience. I have a hobby with a free service in which I don’t owe anyone anything. And I
  understand your words.

  Wrote a lot of words, but erased everything. It was my subjective opinion that will not change anything, and therefore it is unnecessary.

  Now Why I wrote what I wrote. I feel the need to tell people my opinion on the difference in functionality between chan_sip and res_pjsip. They may be important for decision making. I this information was lacking in the past.

  Dmitriy.

  —

• —–Original Message—

• 06.10.2015 1:22, Joshua Colp пишет:

  Something like auth_message_requests:
  http://lists.digium.com/pipermail/asterisk-users/2015-September/287516.html
  (ugg formating)
  In short:
  – GOIP gate (successfully registered as endpoint) send SIP MESSAGE
  – asterisk send registration request
  – nothing. I now understand that the reason may be exactly the same described below.

  Welltech WP589. Beautifully registered using chan_sip and res_pjsip not logged in. Analyzing the exchange of SIP packets I found a single difference: the format of the “nonce” field. When using a longer nonce (pjsip) this phone simply does not respond to the request packet authorization (as do many hardware and software encountering something incomprehensible). The same behavior was on the built-in nokia 95 SIP client.

  Sorry. Not registration, but INVITE. The client software encounters an unfamiliar SDP headers and simply not responding to SIP packets. The specifics of my service is that I don’t know what SIP client is on the other side. What it supports and what not. To give to configure to a user – not the best idea, because often they do not understand what they onoff and why stops working.

  Some of not fixed:
  https://issues.asterisk.org/jira/browse/ASTERISK-25439
  https://issues.asterisk.org/jira/browse/ASTERISK-25435
  https://issues.asterisk.org/jira/browse/ASTERISK-25421
  https://issues.asterisk.org/jira/browse/ASTERISK-25378
  https://issues.asterisk.org/jira/browse/ASTERISK-25279

  Dmitriy.

• Dmitriy Serov wrote:

  I haven’t heard of this or seen it in testing, I don’t think an issue exists for it.

  I’m not sure there’s anything that could be done here…

  They’re in the queue then.

• On the subject of nonce length there is this issue about chan_sip’s nonce length being too short:
  https://issues.asterisk.org/jira/browse/ASTERISK-25062

  Richard