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    <div class="moz-cite-prefix">06.10.2015 1:22, Joshua Colp пишет:<br>
    </div>
    <blockquote cite="mid:5612F822.10706@digium.com" type="cite">On
      15-10-05 05:58 PM, Dmitriy Serov wrote:
      <br>
      <blockquote type="cite">05.10.2015 23:24, Joshua Colp пишет:
        <br>
        <blockquote type="cite">On 15-10-05 05:22 PM, Dmitriy Serov
          wrote:
          <br>
          <blockquote type="cite">Hello. Do I understand correctly that
            the current implementation
            <br>
            res_pjsip does not support ZRTP?
            <br>
<a class="moz-txt-link-freetext" href="http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html">http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html</a>
            <br>
          </blockquote>
          <br>
          ZRTP is not supported in Asterisk itself.
          <br>
          <br>
          <blockquote type="cite">Nothing has changed since 2013? P.S. I
            greatly regret that moved from
            <br>
            chan_sip to res_pjsip. Previously used very much lacking,
            and much of
            <br>
            the promise failed. Dmitriy Serov.
            <br>
          </blockquote>
          <br>
          Any particular examples?
          <br>
          <br>
        </blockquote>
        <br>
        - opus support. Ok... I know the reason why it is not supported
        fully
        <br>
        this codec. But the existing foreign solution works fine with
        chan_sip
        <br>
        and does not work with res_pjsip works.
        <br>
        - endpoint specific ACL
        <br>
        - No support for SIP message without authorization. For this
        reason, the
        <br>
        previously working functionality of sending and receiving SMS
        from
        <br>
        gateway GOIP had to rewrite their internal Protocol.
        <br>
      </blockquote>
      <br>
      Can you clarify what you mean here? There's an anonymous endpoint
      identifier which can be used for anonymous inbound messages
      basically.
      <br>
    </blockquote>
    <br>
    Something like auth_message_requests:
    <a class="moz-txt-link-freetext" href="http://lists.digium.com/pipermail/asterisk-users/2015-September/287516.html">http://lists.digium.com/pipermail/asterisk-users/2015-September/287516.html</a>  
    (ugg formating)<br>
    In short:<br>
    - GOIP gate (successfully registered as endpoint) send SIP MESSAGE<br>
    - asterisk send registration request<br>
    - nothing.<br>
    <span class="translation-chunk" style="margin: 0px; padding: 0px; outline: 0px; color: rgb(34, 34, 34); font-family: Arial, Helvetica, sans-serif; font-size: 16px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: 19.2px; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: pre-wrap; widows: 1; word-spacing: 0px; -webkit-text-stroke-width: 0px; background-color: rgb(255, 255, 255);"></span>I
    now understand that the reason may be exactly the same described
    below.<br>
    <br>
    <blockquote cite="mid:5612F822.10706@digium.com" type="cite">
      <br>
      <blockquote type="cite">- found hardphones and software phones
        that don't accept "long nonce"
        <br>
        and refuse to register when using res_pjsip
        <br>
      </blockquote>
      <br>
      Have you filed an issue with this and details about the
      hardphones+softphones?
      <br>
    </blockquote>
    <br>
    Welltech WP589. Beautifully registered using chan_sip and res_pjsip
    not logged in. <br>
    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).<br>
    The same behavior was on the built-in nokia 95 SIP client.<br>
    <br>
    <blockquote cite="mid:5612F822.10706@digium.com" type="cite">
      <br>
      <blockquote type="cite">- enable icesupport also leads to problems
        of registration and cannot be
        <br>
        "common solution"
        <br>
      </blockquote>
      <br>
      icesupport is only applied to calls, what happens for
      registration?
      <br>
    </blockquote>
    <br>
    Sorry. Not registration, but INVITE.<br>
    The client software encounters an unfamiliar SDP headers and simply
    not responding to SIP packets.<br>
    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. <br>
    To give to configure to a user - not the best idea, because often
    they do not understand what they onoff and why stops working.<br>
    <br>
    <blockquote cite="mid:5612F822.10706@digium.com" type="cite">
      <br>
      <blockquote type="cite">- issue tracker now contains multiple
        error messages that arise every
        <br>
        day and reboot my server (which cannot be called a production)
        <br>
        - And watchdog logs SegFaults and Hangs including other stacks
        that are
        <br>
        not yet documented in the issue tracker.
        <br>
      </blockquote>
      <br>
      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.
      <br>
      <br>
    </blockquote>
    <br>
    Some of not fixed:<br>
    <a class="moz-txt-link-freetext" href="https://issues.asterisk.org/jira/browse/ASTERISK-25439">https://issues.asterisk.org/jira/browse/ASTERISK-25439</a><br>
    <a class="moz-txt-link-freetext" href="https://issues.asterisk.org/jira/browse/ASTERISK-25435">https://issues.asterisk.org/jira/browse/ASTERISK-25435</a><br>
    <a class="moz-txt-link-freetext" href="https://issues.asterisk.org/jira/browse/ASTERISK-25421">https://issues.asterisk.org/jira/browse/ASTERISK-25421</a><br>
    <a class="moz-txt-link-freetext" href="https://issues.asterisk.org/jira/browse/ASTERISK-25378">https://issues.asterisk.org/jira/browse/ASTERISK-25378</a><br>
    <a class="moz-txt-link-freetext" href="https://issues.asterisk.org/jira/browse/ASTERISK-25279">https://issues.asterisk.org/jira/browse/ASTERISK-25279</a><br>
    <br>
    Dmitriy.<br>
    <br>
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