Hello all,
I can't seem to get either RTPProxy or RTPEngine to work correctly. I have decided to concentrate on RTPEngine because I have read that it works the best. I am using Asterisk with Kamailio in front. When I make calls, I see RTPEngine being hit but I continually get the error "Call-ID not found", calls work but media traffic doesn't go through RTPEngine. I have tried many combination of the flags but I seem to always get the same error. When I look at the tcpdump log, I see that the media offer is not on the RTPEngine port. Is there a common mis-configuration error that can cause this?
I know I can send all my logs and configurations but I really want to try and resolve this as a learning experience.
Thanks all, -Steve
Here is my start up => rtpengine --interface 111.121.22.11!27.22.132.10 --listen-ng 127.0.0.1:12221 --dtls-passive -f -m 10000 -M 20000 -E -L 7 --log-facility=local1
My offer and answer => rtpengine_offer("trust-address replace-session-connection replace-origin");
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Wilkins, Steve Sent: Wednesday, August 22, 2018 8:43 AM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: [SR-Users] Struggling with RTPProxy and RTPEngine
Hello all,
I can't seem to get either RTPProxy or RTPEngine to work correctly. I have decided to concentrate on RTPEngine because I have read that it works the best. I am using Asterisk with Kamailio in front. When I make calls, I see RTPEngine being hit but I continually get the error "Call-ID not found", calls work but media traffic doesn't go through RTPEngine. I have tried many combination of the flags but I seem to always get the same error. When I look at the tcpdump log, I see that the media offer is not on the RTPEngine port. Is there a common mis-configuration error that can cause this?
I know I can send all my logs and configurations but I really want to try and resolve this as a learning experience.
Thanks all, -Steve
Do you load the module ?
De : sr-users sr-users-bounces@lists.kamailio.org De la part de Wilkins, Steve Envoyé : mercredi 22 août 2018 14:50 À : Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Objet : Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Here is my start up => rtpengine --interface 111.121.22.11!27.22.132.10 --listen-ng 127.0.0.1:12221 --dtls-passive -f -m 10000 -M 20000 -E -L 7 --log-facility=local1
My offer and answer => rtpengine_offer("trust-address replace-session-connection replace-origin");
From: sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Wilkins, Steve Sent: Wednesday, August 22, 2018 8:43 AM To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Struggling with RTPProxy and RTPEngine
Hello all,
I can't seem to get either RTPProxy or RTPEngine to work correctly. I have decided to concentrate on RTPEngine because I have read that it works the best. I am using Asterisk with Kamailio in front. When I make calls, I see RTPEngine being hit but I continually get the error "Call-ID not found", calls work but media traffic doesn't go through RTPEngine. I have tried many combination of the flags but I seem to always get the same error. When I look at the tcpdump log, I see that the media offer is not on the RTPEngine port. Is there a common mis-configuration error that can cause this?
I know I can send all my logs and configurations but I really want to try and resolve this as a learning experience.
Thanks all, -Steve
Yes I do.
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Nicolas Breuer Sent: Wednesday, August 22, 2018 8:55 AM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Do you load the module ?
De : sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org> De la part de Wilkins, Steve Envoyé : mercredi 22 août 2018 14:50 À : Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org> Objet : Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Here is my start up => rtpengine --interface 111.121.22.11!27.22.132.10 --listen-ng 127.0.0.1:12221 --dtls-passive -f -m 10000 -M 20000 -E -L 7 --log-facility=local1
My offer and answer => rtpengine_offer("trust-address replace-session-connection replace-origin");
From: sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Wilkins, Steve Sent: Wednesday, August 22, 2018 8:43 AM To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Struggling with RTPProxy and RTPEngine
Hello all,
I can't seem to get either RTPProxy or RTPEngine to work correctly. I have decided to concentrate on RTPEngine because I have read that it works the best. I am using Asterisk with Kamailio in front. When I make calls, I see RTPEngine being hit but I continually get the error "Call-ID not found", calls work but media traffic doesn't go through RTPEngine. I have tried many combination of the flags but I seem to always get the same error. When I look at the tcpdump log, I see that the media offer is not on the RTPEngine port. Is there a common mis-configuration error that can cause this?
I know I can send all my logs and configurations but I really want to try and resolve this as a learning experience.
Thanks all, -Steve
I was attempting to log to local1 => rtpengine --interface 172.21.1.108!34.226.187.61 --listen-ng 127.0.0.1:12221 --dtls-passive -f -m 10000 -M 20000 -E -L 7 --log-facility=local1. However, even after adding "local1.* /var/log/rtpengine.log" to /etc/rsyslog.conf and restarting rsyslog, I get no logs. I am on Cento7. I do this for Kamailio and logging works.
Thank you for your response, -Steve
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Richard Fuchs Sent: Wednesday, August 22, 2018 8:57 AM To: sr-users@lists.kamailio.org Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
It may be more helpful to post some logs from rtpengine. You should never see "Call-ID not found" from an offer.
Cheers
On 2018-08-22 08:49, Wilkins, Steve wrote: Here is my start up => rtpengine --interface 111.121.22.11!27.22.132.10 --listen-ng 127.0.0.1:12221 --dtls-passive -f -m 10000 -M 20000 -E -L 7 --log-facility=local1
My offer and answer => rtpengine_offer("trust-address replace-session-connection replace-origin");
From: sr-users sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org On Behalf Of Wilkins, Steve Sent: Wednesday, August 22, 2018 8:43 AM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org Subject: [SR-Users] Struggling with RTPProxy and RTPEngine
Hello all,
I can't seem to get either RTPProxy or RTPEngine to work correctly. I have decided to concentrate on RTPEngine because I have read that it works the best. I am using Asterisk with Kamailio in front. When I make calls, I see RTPEngine being hit but I continually get the error "Call-ID not found", calls work but media traffic doesn't go through RTPEngine. I have tried many combination of the flags but I seem to always get the same error. When I look at the tcpdump log, I see that the media offer is not on the RTPEngine port. Is there a common mis-configuration error that can cause this?
I know I can send all my logs and configurations but I really want to try and resolve this as a learning experience.
Thanks all, -Steve
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
On Wed, Aug 22, 2018 at 12:49:54PM +0000, Wilkins, Steve wrote:
My offer and answer => rtpengine_offer("trust-address replace-session-connection replace-origin");
If this is really your config you should change the offer to answer for the answer part of the config. Either use rtpengine_offer and rtengine_answer in the correct places of simply use rptengine_manage and let the module figure out the right command.
I.O.W. you aren't giving any indepth info on your config.
Is there a particular reason why you'd want to involve an rtp proxy in your setup? Considering Asterisk is facing a public interface, media can be set up to flow direct between UACs and Asterisk.
On Wed, Aug 22, 2018 at 9:57 AM Daniel Tryba d.tryba@pocos.nl wrote:
Hello,
Yes, I don’t the UAC to have to whitelist the Asterisk Server. Everything works great for SIP traffic and I want the same thing for RTP traffic, but I just can’t get rtpengine to work (and I have tried many configurations). From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Sergiu Pojoga Sent: Wednesday, August 22, 2018 11:23 AM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Is there a particular reason why you'd want to involve an rtp proxy in your setup? Considering Asterisk is facing a public interface, media can be set up to flow direct between UACs and Asterisk.
On Wed, Aug 22, 2018 at 9:57 AM Daniel Tryba <d.tryba@pocos.nlmailto:d.tryba@pocos.nl> wrote: On Wed, Aug 22, 2018 at 12:49:54PM +0000, Wilkins, Steve wrote:
My offer and answer => rtpengine_offer("trust-address replace-session-connection replace-origin");
If this is really your config you should change the offer to answer for the answer part of the config. Either use rtpengine_offer and rtengine_answer in the correct places of simply use rptengine_manage and let the module figure out the right command.
I.O.W. you aren't giving any indepth info on your config. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Why would the UAC need to 'whitelist' Asterisk?
Here's an example of an inbound call from Asterisk to a UAC with a sip proxy in the middle. Notice how the proxy mangles SDP in the 200 OK reply. Media will flow directly.
Not saying you shouldn't figure it out how to setup and use rtp proxies, just that it's not a must in all scenarios, especially when you have a media gateway such as Asterisk listening on a public interface.
[image: image.png]
On Wed, Aug 22, 2018 at 12:21 PM Wilkins, Steve swwilkins@mitre.org wrote:
The SIP traffic is working this way for me but I still see RTP traffic going directly from Asterisk to the UAC, which means they need to whitelist asterisk IP. Am I missing something?
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Sergiu Pojoga Sent: Wednesday, August 22, 2018 12:45 PM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Why would the UAC need to 'whitelist' Asterisk?
Here's an example of an inbound call from Asterisk to a UAC with a sip proxy in the middle. Notice how the proxy mangles SDP in the 200 OK reply. Media will flow directly.
Not saying you shouldn't figure it out how to setup and use rtp proxies, just that it's not a must in all scenarios, especially when you have a media gateway such as Asterisk listening on a public interface.
[image.png]
On Wed, Aug 22, 2018 at 12:21 PM Wilkins, Steve <swwilkins@mitre.orgmailto:swwilkins@mitre.org> wrote: Hello,
Yes, I don’t the UAC to have to whitelist the Asterisk Server. Everything works great for SIP traffic and I want the same thing for RTP traffic, but I just can’t get rtpengine to work (and I have tried many configurations). From: sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Sergiu Pojoga Sent: Wednesday, August 22, 2018 11:23 AM To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org> Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Is there a particular reason why you'd want to involve an rtp proxy in your setup? Considering Asterisk is facing a public interface, media can be set up to flow direct between UACs and Asterisk.
On Wed, Aug 22, 2018 at 9:57 AM Daniel Tryba <d.tryba@pocos.nlmailto:d.tryba@pocos.nl> wrote: On Wed, Aug 22, 2018 at 12:49:54PM +0000, Wilkins, Steve wrote:
My offer and answer => rtpengine_offer("trust-address replace-session-connection replace-origin");
If this is really your config you should change the offer to answer for the answer part of the config. Either use rtpengine_offer and rtengine_answer in the correct places of simply use rptengine_manage and let the module figure out the right command.
I.O.W. you aren't giving any indepth info on your config. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Maybe you have enabled direct-media on the Asterisk side?
With best regards
Florian Floimair Innovation - Software-Development
COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.comhttp://www.commend.com/
Security and Communication by Commend
FN 178618z | LG Salzburg
Von: sr-users sr-users-bounces@lists.kamailio.org im Auftrag von "Wilkins, Steve" swwilkins@mitre.org Antworten an: "Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org Datum: Donnerstag, 23. August 2018 um 08:07 An: "Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org Betreff: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
The SIP traffic is working this way for me but I still see RTP traffic going directly from Asterisk to the UAC, which means they need to whitelist asterisk IP. Am I missing something?
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Sergiu Pojoga Sent: Wednesday, August 22, 2018 12:45 PM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Why would the UAC need to 'whitelist' Asterisk?
Here's an example of an inbound call from Asterisk to a UAC with a sip proxy in the middle. Notice how the proxy mangles SDP in the 200 OK reply. Media will flow directly.
Not saying you shouldn't figure it out how to setup and use rtp proxies, just that it's not a must in all scenarios, especially when you have a media gateway such as Asterisk listening on a public interface.
[image.png]
On Wed, Aug 22, 2018 at 12:21 PM Wilkins, Steve <swwilkins@mitre.orgmailto:swwilkins@mitre.org> wrote: Hello,
Yes, I don’t the UAC to have to whitelist the Asterisk Server. Everything works great for SIP traffic and I want the same thing for RTP traffic, but I just can’t get rtpengine to work (and I have tried many configurations). From: sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Sergiu Pojoga Sent: Wednesday, August 22, 2018 11:23 AM To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org> Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Is there a particular reason why you'd want to involve an rtp proxy in your setup? Considering Asterisk is facing a public interface, media can be set up to flow direct between UACs and Asterisk.
On Wed, Aug 22, 2018 at 9:57 AM Daniel Tryba <d.tryba@pocos.nlmailto:d.tryba@pocos.nl> wrote: On Wed, Aug 22, 2018 at 12:49:54PM +0000, Wilkins, Steve wrote:
My offer and answer => rtpengine_offer("trust-address replace-session-connection replace-origin");
If this is really your config you should change the offer to answer for the answer part of the config. Either use rtpengine_offer and rtengine_answer in the correct places of simply use rptengine_manage and let the module figure out the right command.
I.O.W. you aren't giving any indepth info on your config. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi Florian,
Thank you for your response. I checked and direct-media is off. Just as a recap, here is where I am
I just can't get rtpengine to work. I have tried multiple configurations, but to no avail. Note that calls work good if rtpengine is disabled.
Here is my setup =>
Public IP: 20.20.20.20 Private IP 10.10.10.10
flow => webrtc client <-> kamailio+rtpengine <-> asterisk <-> kamailio <-> legacy sip phone
rtpenngine startup (I have tried a few different startups) => rtpengine --interface=int/10.10.10.10 --interface=ext/10.10.10.10!20.20.20.20 --listen-ng=127.0.0.1:12221 --pidfile=/var/run/rtpengine --dtls-passive -f -m 10000 -M 20000 -E
kamailio => Invites: rtpengine_manage("trust-address replace-origin replace-session-connection direction=ext direction=int ICE=remove RTP/AVP"); Reply's: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF");
I have tried direction ext ext; and many other combinations, each producing its own incorrect behavior.
Thanks again, Steve
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Floimair Florian Sent: Thursday, August 23, 2018 3:33 AM To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Maybe you have enabled direct-media on the Asterisk side?
With best regards
Florian Floimair Innovation - Software-Development
COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.comhttp://www.commend.com/
Security and Communication by Commend
FN 178618z | LG Salzburg
Von: sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org> im Auftrag von "Wilkins, Steve" <swwilkins@mitre.orgmailto:swwilkins@mitre.org> Antworten an: "Kamailio (SER) - Users Mailing List" <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org> Datum: Donnerstag, 23. August 2018 um 08:07 An: "Kamailio (SER) - Users Mailing List" <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org> Betreff: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
The SIP traffic is working this way for me but I still see RTP traffic going directly from Asterisk to the UAC, which means they need to whitelist asterisk IP. Am I missing something?
From: sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Sergiu Pojoga Sent: Wednesday, August 22, 2018 12:45 PM To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org> Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Why would the UAC need to 'whitelist' Asterisk?
Here's an example of an inbound call from Asterisk to a UAC with a sip proxy in the middle. Notice how the proxy mangles SDP in the 200 OK reply. Media will flow directly.
Not saying you shouldn't figure it out how to setup and use rtp proxies, just that it's not a must in all scenarios, especially when you have a media gateway such as Asterisk listening on a public interface.
[image.png]
On Wed, Aug 22, 2018 at 12:21 PM Wilkins, Steve <swwilkins@mitre.orgmailto:swwilkins@mitre.org> wrote: Hello,
Yes, I don’t the UAC to have to whitelist the Asterisk Server. Everything works great for SIP traffic and I want the same thing for RTP traffic, but I just can’t get rtpengine to work (and I have tried many configurations). From: sr-users <sr-users-bounces@lists.kamailio.orgmailto:sr-users-bounces@lists.kamailio.org> On Behalf Of Sergiu Pojoga Sent: Wednesday, August 22, 2018 11:23 AM To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org> Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
Is there a particular reason why you'd want to involve an rtp proxy in your setup? Considering Asterisk is facing a public interface, media can be set up to flow direct between UACs and Asterisk.
On Wed, Aug 22, 2018 at 9:57 AM Daniel Tryba <d.tryba@pocos.nlmailto:d.tryba@pocos.nl> wrote: On Wed, Aug 22, 2018 at 12:49:54PM +0000, Wilkins, Steve wrote:
My offer and answer => rtpengine_offer("trust-address replace-session-connection replace-origin");
If this is really your config you should change the offer to answer for the answer part of the config. Either use rtpengine_offer and rtengine_answer in the correct places of simply use rptengine_manage and let the module figure out the right command.
I.O.W. you aren't giving any indepth info on your config. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
On Wed, Aug 22, 2018 at 05:05:02PM +0000, Wilkins, Steve wrote:
The SIP traffic is working this way for me but I still see RTP traffic going directly from Asterisk to the UAC, which means they need to whitelist asterisk IP. Am I missing something?
In what sense do they need whitelisting? In a common NATed solution where is no white/blacklist needed. UA gets RTP endpoints from SDP, starts sending packets to ip/port and the destination will send back packets to the source ip/port, the router/firewall will just send this to the actual UA. I have yet to find an UA that cares about where the RTP stream is coming from with regards to the SIP traffic.
The expectation that the RTP will come from the same place as the signalling does exist in some sclerotic telco interconnects, since big brand SBCs would ordinarily meet this need and big brand SBCs are the only thing sclerotic telcos understand.
But I agree that it's an exotic and uncommon requirement, certainly not applicable to anything like end-user UAs or ordinary business SIP devices.
-- Sent from mobile. Apologies for brevity and errors.
-----Original Message----- From: Daniel Tryba d.tryba@pocos.nl To: "Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org Sent: Thu, 23 Aug 2018 4:35 AM Subject: Re: [SR-Users] Struggling with RTPProxy and RTPEngine
On Wed, Aug 22, 2018 at 05:05:02PM +0000, Wilkins, Steve wrote:
The SIP traffic is working this way for me but I still see RTP traffic going directly from Asterisk to the UAC, which means they need to whitelist asterisk IP. Am I missing something?
In what sense do they need whitelisting? In a common NATed solution where is no white/blacklist needed. UA gets RTP endpoints from SDP, starts sending packets to ip/port and the destination will send back packets to the source ip/port, the router/firewall will just send this to the actual UA. I have yet to find an UA that cares about where the RTP stream is coming from with regards to the SIP traffic.
That's good old sngrep.
Well, okay, it's not old, it's still the new news.
https://github.com/irontec/sngrep
-- Sent from mobile. Apologies for brevity and errors.
-
Alex Balashov -
Daniel Tryba -
Floimair Florian -
Kevin Olbrich -
Nicolas Breuer -
Richard Fuchs -
Sergiu Pojoga -
Wilkins, Steve