[SR-Users] Struggling with RTPProxy and RTPEngine
Wilkins, Steve
swwilkins at mitre.org
Fri Aug 24 14:37:27 CEST 2018
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 at lists.kamailio.org> On Behalf Of Floimair Florian
Sent: Thursday, August 23, 2018 3:33 AM
To: Kamailio (SER) - Users Mailing List <sr-users at 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
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Von: sr-users <sr-users-bounces at lists.kamailio.org<mailto:sr-users-bounces at lists.kamailio.org>> im Auftrag von "Wilkins, Steve" <swwilkins at mitre.org<mailto:swwilkins at mitre.org>>
Antworten an: "Kamailio (SER) - Users Mailing List" <sr-users at lists.kamailio.org<mailto:sr-users at lists.kamailio.org>>
Datum: Donnerstag, 23. August 2018 um 08:07
An: "Kamailio (SER) - Users Mailing List" <sr-users at lists.kamailio.org<mailto:sr-users at 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 at lists.kamailio.org<mailto:sr-users-bounces at lists.kamailio.org>> On Behalf Of Sergiu Pojoga
Sent: Wednesday, August 22, 2018 12:45 PM
To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org<mailto:sr-users at 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 at mitre.org<mailto:swwilkins at 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 at lists.kamailio.org<mailto:sr-users-bounces at lists.kamailio.org>> On Behalf Of Sergiu Pojoga
Sent: Wednesday, August 22, 2018 11:23 AM
To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org<mailto:sr-users at 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 at pocos.nl<mailto:d.tryba at 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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