[asterisk-users] does res_pjsip support ZRTP?

Dmitriy Serov
Tue, 06 Oct 2015 14:48:03 -0000

06.10.2015 1:22, Joshua Colp ?????:
> On 15-10-05 05:58 PM, Dmitriy Serov wrote:
>> 05.10.2015 23:24, Joshua Colp ?????:
>>> 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.

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.

>
>> - 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?

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.

>
>> - enable icesupport also leads to problems of registration and cannot
be
>> "common solution"
>
> icesupport is only applied to calls, what happens for registration?

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.

>
>> - 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.
>

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.

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