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[BryantZ at zktech.com]

I have an asterisk 13 server behind NAT on a dynamic IP Address. It is running the PJSIP Stack
It is registering to another asterisk 13 server that is on a Static IP outside of the firewall at a different location (also on the PJSIP Stack).

How do we implement STUN/ICE on the server behind the dynamic Address. It does not appear to be registering properly without knowing the NAT pubic address. When I manually add external_media_address and external_signaling_address to the pjsipconfig registration seams to work, but knowing that the IP could change really means I need some kind of STUN/ICE similar to what we ran with chan_sip.
I can find limited documentation on this, and what I have found does not show how to set a stun server to make the ice_support field work on an endpoint.

Can anyone advise where I could find an answer to this.

Thanks in advance for any ideas you can offer.

Bryant

[jcolp at digium.com]

Bryant Zimmerman wrote:

[BryantZ at zktech.com]

Joshua

Since there is no automated way currently built in to update the external signaling and media address information.
Does the realtime pjsip support having the transport contexts section being pulled from a database table?
I was thinking a cron script updating the table and forcing a reload each time an IP address changed might a workable solution.

Thanks
Bryant

From: "Joshua Colp" <jcolp@digium.com> Sent: Tuesday, January 26, 2016 7:39 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] PJSIP Stun/ICE
Bryant Zimmerman wrote: > I have an asterisk 13 server behind NAT on a dynamic IP Address. It is > running the PJSIP Stack > It is registering to another asterisk 13 server that is on a Static IP > outside of the firewall at a different location (also on the PJSIP Stack). > How do we implement STUN/ICE on the server behind the dynamic Address. > It does not appear to be registering properly without knowing the NAT > pubic address. When I manually add external_media_address and > external_signaling_address to the pjsipconfig registration seams to > work, but knowing that the IP could change really means I need some kind > of STUN/ICE similar to what we ran with chan_sip. > I can find limited documentation on this, and what I have found does not > show how to set a stun server to make the ice_support field work on an > endpoint. > Can anyone advise where I could find an answer to this. > Thanks in advance for any ideas you can offer. > Bryant The res_pjsip module does not currently support an auto-updating mechanism for the external signaling and media address information. -- 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

[jcolp at digium.com]

Bryant Zimmerman wrote:

[BryantZ at zktech.com]

Joshua

So once a transport is pulled from the transports table in realtime during asterisk startup it can't get any updates?
Can a new transport be added to the table and the associated endpoints be updated to use the new transport, or are transport types only read at startup across the board?

Thanks Bryant

From: "Joshua Colp" <jcolp@digium.com> Sent: Tuesday, January 26, 2016 8:10 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] PJSIP Stun/ICE
Bryant Zimmerman wrote: > Joshua > Since there is no automated way currently built in to update the > external signaling and media address information. > Does the realtime pjsip support having the transport contexts section > being pulled from a database table? > I was thinking a cron script updating the table and forcing a reload > each time an IP address changed might a workable solution. No, once loaded the transports can not be changed. -- 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

[jcolp at digium.com]

Bryant Zimmerman wrote:

[BryantZ at zktech.com]

Daniel

Thank you for your response. I was considering this as well. I have a script that monitors the IP Address now. I was hoping to use the real-time transports table now that alembic creates. I am trying to figure out which pjsip module is responsible for the transports contexts as I need to now configure it in the sorcery.conf file. I thought it would be under the [res_pjsip] context, but it is not even trying to pull from my transports table when it is there. I am hoping someone will know what module it is in so I can move my configuration under the correct context.

Thanks Bryant

From: "Daniel Heckl" <daniel.heckl@gmail.com> Sent: Tuesday, January 26, 2016 10:15 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] PJSIP Stun/ICE
Bryant, I have the same problem with dynamic public IPs and PJSIP. What is your idea to solve the problem? My suggestion would be to write a script that monitors the change, pjsip.transports.conf updated and Asterisk restarts? Daniel > Am 26.01.2016 um 14:21 schrieb Joshua Colp <jcolp@digium.com>: > > Bryant Zimmerman wrote: >> Joshua >> So once a transport is pulled from the transports table in realtime >> during asterisk startup it can't get any updates? >> Can a new transport be added to the table and the associated endpoints >> be updated to use the new transport, or are transport types only read at >> startup across the board? > > Transports can only be loaded at startup. This stems from PJSIP not being dynamic with transports (it doesn't like its environment changed to that degree while in use). I'm afraid if your IP changes you'd have to restart Asterisk when you are using PJSIP. > > -- > 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 -- _____________________________________________________________________ -- 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

[cloos at jhcloos.com]

[jcolp at digium.com]

James Cloos wrote:

[francis.mendoza8 at gm...]

Hi JC,

I have the same case as you are my server has static public IP assigned and my client has public dynamic IP address in order to connect them without issue what I did was to setup openvpn in my other side that has public static IP and then the client server asterisk will connect into it and they will communicate with the VPN local IP adresses that I assigned. hope this 'workaround' helps 


~Cheers

On Wednesday, 27 January 2016, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:

[BryantZ at zktech.com]

Joshua

I look forward to improvements as time goes on with PJSIP.
I have been trying all day to get the Transport objects to pull from a real-time table. The documentation says it is possible, but does not show any examples. I am hoping to have the Transports pulled from the table at asterisk startup and then add additional as necessary. Using reloads to make the new Transports available. I understand the limitation of not being able to change existing and can live with that for now.

Do you know if there is anything special I have to do in the sorcery.conf to make the Transports pull from the real-time side of things. All my other tables are working.

I disagree with the user that things PJSIP is worthless. There are some issues to work out long term, and documentation will get better over time as more of us work with it and contribute back. Thanks for all you have assisted with around PJSIP.

Bryant
From: "Joshua Colp" <jcolp@digium.com> Sent: Tuesday, January 26, 2016 8:40 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] PJSIP Stun/ICE
James Cloos wrote: >>>>>> "JC" == Joshua Colp<jcolp@digium.com> writes: > > JC> This stems from PJSIP not being dynamic with transports (it > JC> doesn't like its environment changed to that degree while > JC> in use). I'm afraid if your IP changes you'd have to restart > JC> Asterisk when you are using PJSIP. > > Wow. > > I say this having voted for pjsip over the listed alternatives back when > the plan to depricate chan_sip was first floated: > > That should have excluded pj from the options. Which of course means > there were no reasonable options. PJSIP doesn't like changing existing transports, the NAT functionality is provided by the Asterisk implementation and can't be reloaded as a side effect due to the heavy handed restriction. With work it could be changed to allow the non low level things to be changed. What you can't do with PJSIP is create a UDP transport, reload, and have it removed. Once it's there it is there unless you restart. > > Can ari get around that bug? ARI is a REST interface to Asterisk, it doesn't have anything to do with this. > > Lack of full support for traversing nat makes pjsip worthless for a > large number of users. And the whole point of realtime is to have all > of the rt config fully dymanic. I disagree that it makes it worthless for a large number of users. It's only within the last few days that a few people have run into this particular issue where they have a public IP address that is changing a lot and PJSIP does not support changing it without a restart. If it were a huge sweeping issue we'd be seeing it more often. If it continues to show up a community member or us (heck maybe even myself in my spare time) may look into implementing it. > > If ari cannot avoid that limitation, chan_sip should get full ongoing > maintainance until pjsip is fixed. The support level for chan_sip has already been changed and was announced long ago. Patches will continue to be accepted for it and community members can support it. We (Digium) are putting our effort towards PJSIP. -- 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

[jcolp at digium.com]

Bryant Zimmerman wrote:

[cloos at jhcloos.com]

[george.joseph at fairv...]

[johnkiniston at gmail.com]

Just an idea for a work around, Have you thought about putting a proxy between your PBX and the Internet such as openSIPS or Kamilio?


That way you may not need to change your IP inside pjsip, Let your proxy handle it.