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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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