22 Aug
2018
22 Aug
'18
2:43 p.m.
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
22 Aug
22 Aug
2:49 p.m.
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(a)lists.kamailio.org> On Behalf Of Wilkins, Steve
Sent: Wednesday, August 22, 2018 8:43 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)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
2:55 p.m.
Do you load the module ?
De : sr-users <sr-users-bounces(a)lists.kamailio.org> De la part de Wilkins, Steve
Envoyé : mercredi 22 août 2018 14:50
À : Kamailio (SER) - Users Mailing List <sr-users(a)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.org<mailto: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<mailto: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
3:48 p.m.
Yes I do.
From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Nicolas Breuer
Sent: Wednesday, August 22, 2018 8:55 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)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.org<mailto: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<mailto: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.org<mailto: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<mailto: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
2:57 p.m.
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(a)lists.kamailio.org> *On Behalf Of
*Wilkins, Steve
*Sent:* Wednesday, August 22, 2018 8:43 AM
*To:* Kamailio (SER) - Users Mailing List <sr-users(a)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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
3:54 p.m.
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(a)lists.kamailio.org> On Behalf Of Richard Fuchs
Sent: Wednesday, August 22, 2018 8:57 AM
To: sr-users(a)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.org><mailto: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><mailto: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.org<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
3:55 p.m.
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.
5:22 p.m.
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(a)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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
6:20 p.m.
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(a)lists.kamailio.org> On Behalf Of Sergiu Pojoga
Sent: Wednesday, August 22, 2018 11:23 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)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.nl<mailto: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.org<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
6:44 p.m.
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(a)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(a)lists.kamailio.org> * On Behalf Of *Sergiu
Pojoga
*Sent:* Wednesday, August 22, 2018 11:23 AM
*To:* Kamailio (SER) - Users Mailing List <sr-users(a)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(a)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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
7:05 p.m.
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(a)lists.kamailio.org> On Behalf Of Sergiu Pojoga
Sent: Wednesday, August 22, 2018 12:45 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)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.org<mailto: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.org<mailto: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<mailto: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.nl<mailto: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.org<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
23 Aug
23 Aug
9:32 a.m.
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.com<http://www.commend.com/>
Security and Communication by Commend
FN 178618z | LG Salzburg
Von: sr-users <sr-users-bounces(a)lists.kamailio.org> im Auftrag von "Wilkins,
Steve" <swwilkins(a)mitre.org>
Antworten an: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.kamailio.org>
Datum: Donnerstag, 23. August 2018 um 08:07
An: "Kamailio (SER) - Users Mailing List" <sr-users(a)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(a)lists.kamailio.org> On Behalf Of Sergiu Pojoga
Sent: Wednesday, August 22, 2018 12:45 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)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.org<mailto: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.org<mailto: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<mailto: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.nl<mailto: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.org<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
24 Aug
24 Aug
2:37 p.m.
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(a)lists.kamailio.org> On Behalf Of Floimair Florian
Sent: Thursday, August 23, 2018 3:33 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)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.com<http://www.commend.com/>
Security and Communication by Commend
FN 178618z | LG Salzburg
Von: sr-users
<sr-users-bounces@lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>>
im Auftrag von "Wilkins, Steve"
<swwilkins@mitre.org<mailto:swwilkins@mitre.org>>
Antworten an: "Kamailio (SER) - Users Mailing List"
<sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>
Datum: Donnerstag, 23. August 2018 um 08:07
An: "Kamailio (SER) - Users Mailing List"
<sr-users@lists.kamailio.org<mailto: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<mailto: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<mailto: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.org<mailto: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.org<mailto: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<mailto: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.nl<mailto: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.
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23 Aug
23 Aug
10:35 a.m.
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.
10:43 a.m.
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(a)pocos.nl>
To: "Kamailio (SER) - Users Mailing List" <sr-users(a)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.
12:45 p.m.
On Thu, Aug 23, 2018 at 04:43:40AM -0400, Alex Balashov wrote:
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.
What a lovely word: sclerotic. First time I've ever seen it :)
But yes, in those cases I'm using a combined kamailio/rtpengine
instance.
10:45 a.m.
Hi,
which debug tool is that in your screenshot?
Thank you very much!
Kevin
Am Mi., 22. Aug. 2018 um 18:47 Uhr schrieb Sergiu Pojoga <pojogas(a)gmail.com
:
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(a)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(a)lists.kamailio.org> * On Behalf Of *Sergiu
Pojoga
*Sent:* Wednesday, August 22, 2018 11:23 AM
*To:* Kamailio (SER) - Users Mailing List <sr-users(a)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(a)pocos.nl> wrote:
On Wed, Aug 22, 2018 at 12:49:54PM +0000, Wilkins, Steve wrote:
_______________________________________________
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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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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10:47 a.m.
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.
2073
days inactive
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17 comments
8 participants
participants (8)
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Alex Balashov -
Daniel Tryba -
Floimair Florian -
Kevin Olbrich -
Nicolas Breuer -
Richard Fuchs -
Sergiu Pojoga -
Wilkins, Steve