VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
serov.d.p at gmail.com
Guest
|
 Posted: Mon Oct 05, 2015 3:22 pm Post subject: [asterisk-users] does res_pjsip support ZRTP? |
 |
|
| 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 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. |
|
| Back to top |
|
 |
jcolp at digium.com
Guest
|
| Posted: Mon Oct 05, 2015 3:24 pm Post subject: [asterisk-users] does res_pjsip support ZRTP? |
|
|
On 15-10-05 05:22 PM, Dmitriy Serov wrote:
ZRTP is not supported in Asterisk itself.
| Quote: | 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?
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
serov.d.p at gmail.com
Guest
|
| Posted: Mon Oct 05, 2015 3:59 pm Post subject: [asterisk-users] does res_pjsip support ZRTP? |
|
|
05.10.2015 23:24, Joshua Colp пишет:
| Quote: | On 15-10-05 05:22 PM, Dmitriy Serov wrote:
ZRTP is not supported in Asterisk itself.
| Quote: | 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.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
jcolp at digium.com
Guest
|
| Posted: Mon Oct 05, 2015 5:22 pm Post subject: [asterisk-users] does res_pjsip support ZRTP? |
|
|
On 15-10-05 05:58 PM, Dmitriy Serov wrote:
| Quote: | 05.10.2015 23:24, Joshua Colp пишет:
| Quote: | On 15-10-05 05:22 PM, Dmitriy Serov wrote:
ZRTP is not supported in Asterisk itself.
| Quote: | 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.
| Quote: | - 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?
| Quote: | - enable icesupport also leads to problems of registration and cannot be
"common solution"
|
icesupport is only applied to calls, what happens for registration?
| Quote: | - 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.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
mjordan at digium.com
Guest
|
| Posted: Tue Oct 06, 2015 8:08 am Post subject: [asterisk-users] does res_pjsip support ZRTP? |
|
|
On Mon, Oct 5, 2015 at 3:58 PM, Dmitriy Serov <serov.d.p@gmail.com> wrote:
| Quote: | 05.10.2015 23:24, Joshua Colp пишет:
| Quote: |
On 15-10-05 05:22 PM, Dmitriy Serov wrote:
ZRTP is not supported in Asterisk itself.
| Quote: | 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
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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
serov.d.p at gmail.com
Guest
|
| Posted: Tue Oct 06, 2015 9:06 am Post subject: [asterisk-users] does res_pjsip support ZRTP? |
|
|
06.10.2015 16:08, Matthew Jordan пишет:
| Quote: | 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
|
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.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
RyanT at OscarWinski.com
Guest
|
| Posted: Tue Oct 06, 2015 9:20 am Post subject: [asterisk-users] does res_pjsip support ZRTP? |
|
|
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dmitriy Serov
Sent: Tuesday, October 06, 2015 10:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] does res_pjsip support ZRTP?
06.10.2015 16:08, Matthew Jordan пишет:
| Quote: | 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
|
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.
--
Personally I find PJSIP many times more difficult, but I'm trying to give time to my learning curve. I see it can be much better to have all the stuff in the database for me as a developer also. Just my two cents.
Travis
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
serov.d.p at gmail.com
Guest
|
| Posted: Tue Oct 06, 2015 9:48 am Post subject: [asterisk-users] does res_pjsip support ZRTP? |
|
|
06.10.2015 1:22, Joshua Colp пишет:
| Quote: | On 15-10-05 05:58 PM, Dmitriy Serov wrote:
| Quote: | 05.10.2015 23:24, Joshua Colp пишет:
| Quote: | On 15-10-05 05:22 PM, Dmitriy Serov wrote:
ZRTP is not supported in Asterisk itself.
| Quote: | 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.
| Quote: |
| Quote: | - 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.
| Quote: |
| Quote: | - 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.
| Quote: |
| Quote: | - 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. |
|
| Back to top |
|
|
jcolp at digium.com
Guest
|
| Posted: Tue Oct 06, 2015 10:30 am Post subject: [asterisk-users] does res_pjsip support ZRTP? |
|
|
Dmitriy Serov wrote:
<snip>
| Quote: | | Quote: |
| Quote: | - 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.
|
I haven't heard of this or seen it in testing, I don't think an issue
exists for it.
| Quote: |
| Quote: |
| Quote: | - 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.
|
I'm not sure there's anything that could be done here...
They're in the queue then.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
rmudgett at digium.com
Guest
|
| Posted: Tue Oct 06, 2015 10:55 am Post subject: [asterisk-users] does res_pjsip support ZRTP? |
|
|
On Tue, Oct 6, 2015 at 10:27 AM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:
| Quote: | Dmitriy Serov wrote:
<snip>
| Quote: | | Quote: |
| Quote: | - 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.
|
I haven't heard of this or seen it in testing, I don't think an issue exists for it.
|
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 |
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
