HOW TO USE RTP PROXY WITH WEBRTC TO AVOID THE USE OF STUN and TURN SERVER
Abhisek Panda
Iñaki Baz Castillo
> For an auido call from a google chorme to google chrome or mozilla to
> mozilla many servers are coming in picture. Like NAT (Both Static /
> Symmetric NAT) , STUN server , ICE , TURN Server .
STUN and TURN server may be the same (a TURN server is also a STUN
server). So just one server.
> By using RTP Proxy can the use of NAT , STUN .,TURN be avoided ??
on that) you want to force the media through a server yes or yes.
> How to use RTP proxy for this.
> What will be the call flow here by using RTP Proxy
> i've these below
> client A , clent B , Sip Server.
there is no a "universal RTP Proxy".
--
Iñaki Baz Castillo
<i...@aliax.net>
Abhisek Panda
Iñaki Baz Castillo
> Hi Castillo ,
> Thanks for the reply.
> I've tried using TURN server earlier , by using which i'm able to
> do audio call from one client to other but it is always not working because
> of bandwidth of turn server i guess. When ever ports are free then only the
> audio call is working it seems. On the other hand i've used open source turn
> server from numb/viagenie (numb.viagenie.ca/) using this turn server is a
> risk as my audio data is not secure for going in the public turn server.
> Hence i'm avoiding to use turn server , instead want to use RTP Proxy which
> can be helpful to avoid stun/turn.
> I've found RTP Proxy below,
> git clone git://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy
>
> Which we have compiled & RTP Proxy is ready as well. So now i want to use
> this RTP proxy with WEBRTC . I'm using Huawei SIp Server.
> I want to know how to use the RTP Proxy to avoide turn & stun server.
> What should be the call flow & how RTP Proxy will be able to avoid TURN /
> STUN uses & can help for flowing RTP between end points.
Calvin Walton
On Tue, 2015-08-18 at 05:31 -0700, Abhisek Panda wrote:
> Hi Castillo ,
> Thanks for the reply.
> I've tried using TURN server earlier , by using which i'm
> able
> to do audio call from one client to other but it is always not
> working
> because of bandwidth of turn server i guess. When ever ports are free
> then
> only the audio call is working it seems. On the other hand i've used
> open
> source turn server from numb/viagenie (numb.viagenie.ca/) using this
> turn
> server is a risk as my audio data is not secure for going in the
> public
> turn server.
> Hence i'm avoiding to use turn server , instead want to use RTP Proxy
> which
> can be helpful to avoid stun/turn.
DTLS) end-to-end, so the TURN server can't see the audio packet
contents.
This relies on the fingerprints in the signalling path being sent over
a secure connection (verified HTTPS/WSS connection, for example).
> I've found RTP Proxy below,
> git clone git://sippy.git.sourceforge.net/gitroot/sippy/rtpproxy
>
> Which we have compiled & RTP Proxy is ready as well. So now i want to
> use
> this RTP proxy with WEBRTC . I'm using Huawei SIp Server.
> I want to know how to use the RTP Proxy to avoide turn & stun server.
> What should be the call flow & how RTP Proxy will be able to avoid
> TURN /
> STUN uses & can help for flowing RTP between end points.
server.
https://github.com/coturn/rfc5766-turn-server/
and
https://github.com/coturn/coturn
are some well-known open-source TURN servers that you can run yourself,
there may be others as well.
Setting up an RTP proxy designed for SIP would be very difficult, sinceWebRTC does not natively use SIP for signalling.
--
Calvin Walton <calvin...@kepstin.ca>
Abhisek Panda
https://github.com/coturn/rfc5766-turn-server/
2. Deployed in a local server (172.xx.xx.51) inside root path
(Not Sure if turn server should be placed before firewall/NAT or after firewall/NAT -- please conform this )
3. While running ./configure found an issue as ,
ERROR: Libevent2 development libraries are not installed properly in required location
I've checked for the issue, which said Libevent2 by default present in Fedora OS but not in Cent OS , which I am using.
For Cent OS Libevent2 need to be installed manually.
4. Manual Installation i did by downloading libevent-2.0.22-stable.tar.gz from http://libevent.org/
and installing in the path /usr/ports/devel/libevent2/
I've run ./configure -- > Make --> Make Install .
libevent2 installed properly without any error.
5. Again tried to run ./configure of Turnserver, Which was error free but 1 warning was there
HIREDIS DEVELOPMENT LIBRARY (libhiredis.*) AND/OR HEADERS (hiredis/*.h)
ARE NOT INSTALLED PROPERLY ON THIS SYSTEM.
THAT'S OK BUT THE TURN SERVER IS BUILDING WITHOUT REDIS SUPPORT.
This message i have ignored as i'm not sure if it will be an issue & installed Turn Server by Make & Make Install Command.
Turn Server Got Installed .
6. Created a configuration file ,
/etc/turnserver.conf & added the below , with a random port 5555. (not sure if this way it will work)
#setting static accounts
# Remember, "static" accounts are not dynamically checked by the turn server process.
user=username:password
# listen ports
listening-port=5555
listening-ip=172.16.6.51
7. I've tried to Run The TURN Server with the below command
turnserver -v -r 172.16.6.51:5555 -a -b turnuserdb.conf -c turnserver.conf -u custom-username -r 172.16.6.51:5555 -p custom-password
With this i was able to make the turn server up.
[root@localhost rfc5766-turn-server-3.2]# ps -el | grep turn
0 S 0 12147 32123 0 80 0 - 236237 epoll_ pts/2 00:00:02 turnserver
8. I've modified the script (call.htm) of my SIPML5 ( like http://sipml5.org/call.htm?svn=230 this page i've my own designed page) client page in this way
// **** Begin Abhisek Modified for using own TURN Server ****
//window.localStorage.setItem('org.doubango.expert.ice_servers', "[{ url: 'stun:stun.l.google.com:19302'}]");
window.localStorage.setItem('org.doubango.expert.ice_servers', "[{ url: 'turn:172.16.6.51:55555'}]");
// **** End Abhisek Modified for using own TURN Server ****
Can you guys give some more suggestion if this all way is proper ??? Or do i need to have some more different setup for having my own TURn server ??
Many Thanks ,
Abhisek