similar to: Doc for PJSIP ICE support ?

Thanks Joshua for replying ! What would you advise : - leaving STUN address empty, in rtp.conf, as "STUN is not required for ICE" - configure it with one public STUN (I'm using stun.voip.ovh.net for this but I don't know how this server really works) Cheers Le mar. 27 oct. 2020 à 09:53, Joshua C. Colp <jcolp at sangoma.com> a écrit : > On Tue, Oct 27, 2020 at 5:35 AM

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

On Tue, Oct 27, 2020 at 5:35 AM Olivier <oza.4h07 at gmail.com> wrote: > Hello, > > Where can I find doc about PJSIP's ice_support parameter ? > > Do you need to configure things elsewhere in Asterisk config files > (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ? > It needs to also be enabled in rtp.conf. > I'm asking because, if

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

On 11/15/17 11:10 AM, Joshua Colp wrote: > On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote: >> On 11/14/17 5:23 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: >>>> Trace with 3 clients. We can hear each other but no video. >>>> >>>>

>>>>> "AS" == A J Stiles <asterisk_list at earthshod.co.uk> writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordable. -JimC -- James Cloos <cloos at

Le lun. 26 août 2019 à 12:07, Joshua C. Colp <jcolp at digium.com> a écrit : > ... > > libpjnath is the ICE/STUN/TURN library which is used by res_rtp_asterisk > for that functionality. If you're using WebRTC or ICE/STUN/TURN, then you > would be using that library. > Yes, I'm using ICE/STUN/TURN. That explains libpjnath usage. Thank you sharing this here. Now

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 at digium.com>: > > Bryant Zimmerman wrote: >> Joshua >> So once a transport is

I have many endpoints and each endpoint has some parameter in common so i wonder is there any way to config one for all endpoints? Like in my example I have two endpoints and I repeat the same thing, [100] type=endpoint aors=100 auth=100-auth allow=ulaw,alaw,gsm,g726 context=from-internal callerid=device <100> dtmf_mode=rfc4733 use_avpf=no ice_support=no

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

Hello, I've got an Asterisk 11.13.1 system running on a Debian Jessie platform. This system's extensions.conf doesn't include any reference to PJSIP, yet (only using chan_sip at the moment). This morning, it failed with: Aug 26 09:07:33 foobar kernel: [6534231.776418] asterisk[9701]: segfault at 3c ip 00007f02f5d9a7dd sp 00007f02f0b0b260 error 4 in libpjnath.so.2[7f02f5d84000+26000]

how if I use the auto generate once from freepbx ? On Tue, Sep 22, 2015 at 10:12 PM, Ishfaq Malik <ish at pack-net.co.uk> wrote: > > > On 22 September 2015 at 16:04, Thyda ENG <engthyda at gmail.com> wrote: > >> I have many endpoints and each endpoint has some parameter in common so i >> wonder is there any way to config one for all endpoints? Like in my

Hello, How should I configure Asterisk (13.7.0) to get persistently PJSIP SIP messages in a log file and not in console ? I would expect adding "debug=yes" in pjsip.conf to produce the same output as "pjsip set logger on". Am I understanding correctly ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is

>>>>> "JC" == Joshua Colp <jcolp at 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

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

Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump on asterisk server showing UDP packet bound for my remote endpoints internal IP: 17:07:57.130212 IP

So, the only thing that is needed in the endpoint definition in pjsip.conf (there is no such file pjsip.endpoint_custom.conf) is *message_context=astsms* Is that correct? Anything I need to do in extensions.conf? I see that the messages are received at Asterisk (when I turn on pjsip set logger on) but they are not delivered to the other endpoint. What gives? Any help appreciated. Thanks! On

> Joshua Colp wrote: > > There should be nothing different, except for how you configure things. > What is the full PJSIP configuration? What is the environment where > Asterisk is running? Is ICE actually in use on the other side? What is > the full SIP trace? > The full configuration is here: http://pastebin.com/XqZG1m5X I am connection over TLS / SRTP on port 5063. When

thank you very much. this is exactly whats needed for debug example output for your info [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         icess0x7f5d44081e88 .Added new remote candidate from the request: 2.2.2.2:57536 [Dec 12 15:39:19] DEBUG[2182][C-00000000]: pjproject: <?>:         icess0x7f5d44081e88 .New triggered check added: 1 [Dec 12 15:39:19]