Quantcast

FreeSwitch Proxy + RTPProxy Media server

classic Classic list List threaded Threaded
17 messages Options
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

FreeSwitch Proxy + RTPProxy Media server

adahary
I would like to integrate FS with RTPproxy like openSips and kamailio are well integrated with it.
FS should handle the SIP signaling and the RTPproxy should relay the RTP stream from A to B:
A.sip <=> FS <=>  B.sip  
FS = PASS-THRU
A.rtp <=> RTPproxy <=> B.rtp.

I understand that FS should ask the RTPproxy to allocate UDP ports for both endpoint and then pass-thru-bridge them to cummunicate directly through the RTPproxy.

What I cannot yet figure out is how replace both A and B ip/port sets in SDP with the RTPproxy ip/port-udp.
I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but couldn't figure it out.

how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when when A calls B?
Tried the following with no help.
<action application="set">
<![CDATA[switch_r_sdp=(sdp A with new ip/port)
]]>

</action>
<action application="set">
<![CDATA[switch_m_sdp=(sdp B with new ip/port)
]]>

</action>

Any tip/help/advise will be appriciated.

Regards

Assaf
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

Dmitry Sytchev
What are you trying to achieve with this setup? Centralized nat handling or something else?
My thought was that RTPproxy/mediaproxy has to be externally controlled by own protocol to maintain rtp sessions, how do you plan to do that?


2013/5/23 adahary <[hidden email]>
I would like to integrate FS with RTPproxy like openSips and kamailio are
well integrated with it.
FS should handle the SIP signaling and the RTPproxy should relay the RTP
stream from A to B:
A.sip <=> FS <=>  B.sip
FS = PASS-THRU
A.rtp <=> RTPproxy <=> B.rtp.

I understand that FS should ask the RTPproxy to allocate UDP ports for both
endpoint and then pass-thru-bridge them to cummunicate directly through the
RTPproxy.

What I cannot yet figure out is how replace both A and B ip/port sets in SDP
with the RTPproxy ip/port-udp.
I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but couldn't
figure it out.

how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when when
A calls B?
Tried the following with no help.
<action application="set">
</action>
<action application="set">
</action>

Any tip/help/advise will be appriciated.

Regards

Assaf




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
[hidden email]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Best regards,

Dmitry Sytchev,
IT Engineer

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
[hidden email]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

adahary
My plan is to use the FS as a centralized SIP proxy/registrar and several RTPproxy media server in different locations so users could get better latency when bridged over.

Yes, the RTPproxy is controled by some API commands (google around).
I'll use the xml_curl with PHP to request from the RTPproxy ip/udp ports  and then update A&B SDPs.

I'm just missing the last part of 'update A&B SDPs' ;

adahary
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

jay binks
So why not just <action application="redirect" data="[hidden email] "/> to a closer FS node ??

dont get me wrong, I can see some of the appeal for what your suggesting.
I have also had similar thoughts in the past...
but I keep coming back to the fact you could use FS instead of RTP Proxy... and just sip redirect it.

The ONLY downside I could ever see would be... what if the customer dosnt follow the redirect.
A re-invite with new IP and port is probably a little safer.

Jay





On 23 May 2013 06:59, adahary <[hidden email]> wrote:
My plan is to use the FS as a centralized SIP proxy/registrar and several
RTPproxy media server in different locations so users could get better
latency when bridged over.

Yes, the RTPproxy is controled by some API commands (google around).
I'll use the xml_curl with PHP to request from the RTPproxy ip/udp ports
and then update A&B SDPs.

I'm just missing the last part of 'update A&B SDPs' ;

adahary



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972p7590974.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
[hidden email]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Sincerely

Jay

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
[hidden email]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

Ken Rice-2
In reply to this post by adahary
Why not just use the front end FS box in bypass media mode, then send the
calls to the regional RTP proxies... The sip will act as a controller for
those remote locations...

Or Use OpenSIPs or Kamilio to handle the front end sip, its designed to use
things like RTP Proxy FreeSwitch is not


On 5/22/13 3:59 PM, "adahary" <[hidden email]> wrote:

> My plan is to use the FS as a centralized SIP proxy/registrar and several
> RTPproxy media server in different locations so users could get better
> latency when bridged over.
>
> Yes, the RTPproxy is controled by some API commands (google around).
> I'll use the xml_curl with PHP to request from the RTPproxy ip/udp ports
> and then update A&B SDPs.
>
> I'm just missing the last part of 'update A&B SDPs' ;
>
> adahary
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-
> server-tp7590972p7590974.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> [hidden email]
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> [hidden email]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
[hidden email]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

adahary
In reply to this post by jay binks
Jay,

I need that the RTP will flow directly between the endpoints or through a CLEAR relay server. Redirecting SIP won't do it - it is just another new SIP session which will not work PASS-Thru because of the NAT issue.

adahary
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

adahary
In reply to this post by Ken Rice-2
Ken,

That's exactly what I'm trying to do! FS bypass media and send to RTPproxy to relay the RTP between the two endpoints.

What I'm asking is how to setup FS dialplan to replace the ip/port in both SDP endpoints with the RTPproxy ip/port (given that I already know the RTPproxy ip/port).

thanks

adahary




Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

Ken Rice-2
You miss my point just use FreeSWITCH instead of RTPproxy...


On 5/23/13 12:16 AM, "adahary" <[hidden email]> wrote:

> Ken,
>
> That's exactly what I'm trying to do! FS bypass media and send to RTPproxy
> to relay the RTP between the two endpoints.
>
> What I'm asking is how to setup FS dialplan to replace the ip/port in both
> SDP endpoints with the RTPproxy ip/port (given that I already know the
> RTPproxy ip/port).
>
> thanks
>
> adahary
>
>
>
>
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-
> server-tp7590972p7590983.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> [hidden email]
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> [hidden email]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
[hidden email]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

adahary
Ok, how to setup the controller FS (bypass mode) to relay both endpoints RTP through another FS box (acting like RTP proxy = rtpproxy.fs.com)?
something like dialplan 'bridge endpointB@rtpproxy.fs.com' ?
would that make just the RTP relay between endpoints without initiating a new SIP session with rtpproxy.fs.com?
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

Ken Rice-2
Where you have the following

ENDPOINT A <-> FS A <-> FS B <-> FS C <-> ENDPOINT B

This is a very lose run down, but FS A and C are in bypass media mode, this
causes FS A to simply copy the SDPs from Endpoint A and FS B across the
bridge so its not in the media path.   Same thing happens on FS C...

Leaving media path to go from Endpoint A -> FS B -> Endpoint B

Using some tricks with say custom sip headers (sip_h_X- chan vars see the
wiki) you can instruct FS B where to send the call from there for final
delivery...







On 5/23/13 1:00 AM, "adahary" <[hidden email]> wrote:

> Ok, how to setup the controller FS (bypass mode) to relay both endpoints RTP
> through another FS box (acting like RTP proxy = rtpproxy.fs.com)?
> something like dialplan 'bridge [hidden email]' ?
> would that make just the RTP relay between endpoints without initiating a
> new SIP session with rtpproxy.fs.com?
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-
> server-tp7590972p7590986.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> [hidden email]
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> [hidden email]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
[hidden email]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

adahary
Are FS.A and FS.C logically the same FS box which where both ENDPOINTs A&B registered to?

ENDPOINT A <-> FS A <-> ENDPOINT B   (sip registration)
after FS A bypass mode and 'bridge  ENDPOINT.B@FS.B'
ENDPOINT.A <-> FS B <-> ENDPOINT.B (rtp relay)
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

Ken Rice-2
They may or may not be the same box.


On 5/23/13 1:52 AM, "adahary" <[hidden email]> wrote:

> Are FS.A and FS.C logically the same FS box which where both ENDPOINTs A&B
> registered to?
>
> ENDPOINT A <-> FS A <-> ENDPOINT B   (sip registration)
> after FS A bypass mode and 'bridge  [hidden email]'
> ENDPOINT.A <-> FS B <-> ENDPOINT.B (rtp relay)
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-
> server-tp7590972p7590988.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> [hidden email]
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> [hidden email]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
[hidden email]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

adahary
Ok, this is what I'm about to setup for testing your solution:
1. registering ENDPOINT.1 & 2 (both behind NAT) with FS.A setup to bypass mode.
FS.A has dialplan of ENDPOINT.A@FS.B
2. Setup FS.B seperatly (not in the SIP/media path of  ENDPOINT.1 & 2)
3. Calling from ENDPOINT.1 to ENDPOINT.2
4. The Call hits the FS.A dialplan and being bridged over to FS.B (ENDPOINT.A@FS.B)
Now, on FS.B I should see (wireshark) only RTP UDP stream between ENDPOINT.1 & 2 with NO SIP (re-direct, re-invite).
Is it right?

Please be noticed that:
- Both ENDPOINT.1 & 2 are beind NAT
- ZRTP may be supported for end2end (another reason to have clear RTP relay).

Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

adahary
Ken,

It does not work for me. please check the log below.
ENDPOINT.1 = 1001
ENDPOINT.2 = 1000

2013-05-23 13:10:01.984411 [INFO] mod_dialplan_xml.c:558 Processing 1001 <1001>->1000 in context confitele_endpoints
2013-05-23 13:10:02.004372 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/42341cdc-9982-4845-88be-0576fbd60ea7.tmp.xml
Dialplan: sofia/endpoint-nat/1001@fs-a.com parsing [confitele_endpoints->call_inbound] continue=false
Dialplan: sofia/endpoint-nat/1001@fs-a.com Regex (PASS) [call_inbound] destination_number(1000) =~ /^(\d+)$/ break=on-false
Dialplan: sofia/endpoint-nat/1001@fs-a.com Action set(bypass_media=true) INLINE
EXECUTE sofia/endpoint-nat/1001@fs-a.com set(bypass_media=true)
2013-05-23 13:10:02.004372 [DEBUG] mod_dptools.c:1373 sofia/endpoint-nat/1001@fs-a.com SET [bypass_media]=[true]
Dialplan: sofia/endpoint-nat/1001@fs-a.com Action bridge(${destination_number}@fs-b.com)
2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:167 (sofia/endpoint-nat/1001@fs-a.com) State Change CS_ROUTING -> CS_EXECUTE
2013-05-23 13:10:02.004372 [DEBUG] switch_core_session.c:1340 Send signal sofia/endpoint-nat/1001@fs-a.com [BREAK]
2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:470 (sofia/endpoint-nat/1001@fs-a.com) State ROUTING going to sleep
2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:415 (sofia/endpoint-nat/1001@fs-a.com) Running State Change CS_EXECUTE
2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:477 (sofia/endpoint-nat/1001@fs-a.com) State EXECUTE
2013-05-23 13:10:02.004372 [DEBUG] mod_sofia.c:230 sofia/endpoint-nat/1001@fs-a.com SOFIA EXECUTE
2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:209 sofia/endpoint-nat/1001@fs-a.com Standard EXECUTE
EXECUTE sofia/endpoint-nat/1001@fs-a.com bridge(1000@fs-b.com)
2013-05-23 13:10:02.004372 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables
2013-05-23 13:10:02.004372 [ERR] switch_core_session.c:496 Could not locate channel type 1000@fs-b.com
2013-05-23 13:10:02.004372 [NOTICE] switch_ivr_originate.c:2649 Cannot create outgoing channel of type [1000@fs-b.com] cause: [CHAN_NOT_IMPLEMENTED]
2013-05-23 13:10:02.004372 [DEBUG] switch_ivr_originate.c:3615 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
2013-05-23 13:10:02.004372 [INFO] mod_dptools.c:3098 Originate Failed.  Cause: CHAN_NOT_IMPLEMENTED
2013-05-23 13:10:02.004372 [NOTICE] mod_dptools.c:3218 Hangup sofia/endpoint-nat/1001@fs-a.com [CS_EXECUTE] [CHAN_NOT_IMPLEMENTED]
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

Ken Rice-2
This will absolutely work. You just have to set up the dialplans properly
for the calls to be routed...


On 5/23/13 4:13 AM, "adahary" <[hidden email]> wrote:

> Ken,
>
> It does not work for me. please check the log below.
> ENDPOINT.1 = 1001
> ENDPOINT.2 = 1000
>
> 2013-05-23 13:10:01.984411 [INFO] mod_dialplan_xml.c:558 Processing 1001
> <1001>->1000 in context confitele_endpoints
> 2013-05-23 13:10:02.004372 [CONSOLE] mod_xml_curl.c:318 XML response is in
> /tmp/42341cdc-9982-4845-88be-0576fbd60ea7.tmp.xml
> Dialplan: sofia/endpoint-nat/[hidden email] parsing
> [confitele_endpoints->call_inbound] continue=false
> Dialplan: sofia/endpoint-nat/[hidden email] Regex (PASS) [call_inbound]
> destination_number(1000) =~ /^(\d+)$/ break=on-false
> Dialplan: sofia/endpoint-nat/[hidden email] Action set(bypass_media=true)
> INLINE
> EXECUTE sofia/endpoint-nat/[hidden email] set(bypass_media=true)
> 2013-05-23 13:10:02.004372 [DEBUG] mod_dptools.c:1373
> sofia/endpoint-nat/[hidden email] SET [bypass_media]=[true]
> Dialplan: sofia/endpoint-nat/[hidden email] Action
> bridge(${destination_number}@fs-b.com)
> 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:167
> (sofia/endpoint-nat/[hidden email]) State Change CS_ROUTING -> CS_EXECUTE
> 2013-05-23 13:10:02.004372 [DEBUG] switch_core_session.c:1340 Send signal
> sofia/endpoint-nat/[hidden email] [BREAK]
> 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:470
> (sofia/endpoint-nat/[hidden email]) State ROUTING going to sleep
> 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:415
> (sofia/endpoint-nat/[hidden email]) Running State Change CS_EXECUTE
> 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:477
> (sofia/endpoint-nat/[hidden email]) State EXECUTE
> 2013-05-23 13:10:02.004372 [DEBUG] mod_sofia.c:230
> sofia/endpoint-nat/[hidden email] SOFIA EXECUTE
> 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:209
> sofia/endpoint-nat/[hidden email] Standard EXECUTE
> EXECUTE sofia/endpoint-nat/[hidden email] bridge([hidden email])
> 2013-05-23 13:10:02.004372 [DEBUG] switch_ivr_originate.c:2039 Parsing
> global variables
> 2013-05-23 13:10:02.004372 [ERR] switch_core_session.c:496 Could not locate
> channel type [hidden email]
> 2013-05-23 13:10:02.004372 [NOTICE] switch_ivr_originate.c:2649 Cannot
> create outgoing channel of type [[hidden email]] cause:
> [CHAN_NOT_IMPLEMENTED]
> 2013-05-23 13:10:02.004372 [DEBUG] switch_ivr_originate.c:3615 Originate
> Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED]
> 2013-05-23 13:10:02.004372 [INFO] mod_dptools.c:3098 Originate Failed.
> Cause: CHAN_NOT_IMPLEMENTED
> 2013-05-23 13:10:02.004372 [NOTICE] mod_dptools.c:3218 Hangup
> sofia/endpoint-nat/[hidden email] [CS_EXECUTE] [CHAN_NOT_IMPLEMENTED]
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-
> server-tp7590972p7590996.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> [hidden email]
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> [hidden email]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
[hidden email]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

adahary
Ken,

1000 & 1001 both registered with -> FS.A  (actually all endpoints are registered with FS.A).
FS.B is the media server where I want 1000&1001 RTP to pass through it.

when 1000 calls 1001, then FS.A diaplan sets bypass_media and bridges it to FS.B (sofia/external/1001@FS.B or alternativaly sofia/gateway/fsb/1001) which accepts the call from 1000.

Now, FS.B can simply route the call to 1001@FS.A but that will reroute the RTP through FS.A which is not my goal. I need also 1001 to be somehow 'bypass' to FS.B.

How can I get 1000 RTP -> FS.B -> RTP 1001 (without FS.A in between)?

Regards

Assaf


 
Reply | Threaded
Open this post in threaded view
|  
Report Content as Inappropriate

Re: FreeSwitch Proxy + RTPProxy Media server

samson107
This post has NOT been accepted by the mailing list yet.
In reply to this post by Ken Rice-2
hi all,

It does work for me.

SIP: C1(2001) <--> FS1(PBX) <--> FS2(RTP) <--> C2(2002)
RTP: C1(2001) <--> FS2(RTP) <--> C2(2002)

FS1: 192.168.10.150, internal port 5090, external port 5091
FS2: 192.168.10.157, internal port 5060, external port 5080

1. Setting FS1(PBX)
  $ cd /usr/local/freeswitch/conf/dialplan
  $ vi default.xml
# add this setting
<extension name="phone to FS2 server">
       <condition field="destination_number" expression="^([2].*)$">
         <action application="set" data="bypass_media_after_bridge=true"/>
         <action application="bridge" data="sofia/external/sip:${destination_number}@192.168.10.157:5080" />
  ​      </condition>
  ​    </extension>

  $ vi public.xml
# add this setting
<extension name="Local_Extension">
  ​  <condition field="destination_number" expression="^([2]0[01][0-9])$">
  ​    <action application="set" data="bypass_media_after_bridge=true"/>
  ​    <action application="export" data="dialed_extension=$1"/>
  ​    <action application="set" data="ringback=${us-ring}"/>
  ​    <action application="set" data="transfer_ringback=$${hold_music}"/>
  ​    <action application="set" data="call_timeout=30"/>
      <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
  ​    <action application="answer"/>
  ​    <action application="sleep" data="1000"/>
    ​  <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
  ​  </condition>
</extension>

2. Setting FS2(RTP)
  $ cd /usr/local/freeswitch/conf/dialplan
  $ vi public.xml
# add this setting
  <extension name="transfrom_call">
  ​  <condition field="destination_number" expression="^[2].*$">
  ​    <action application="set" data="bypass_media=false" />
  ​    <action application="set" data="disable-transcoding=true"/>
  ​    <action application="bridge" data="sofia/external/sip:${destination_number}@192.168.10.150:5091" />
  ​  </condition>
  ​</extension>
Loading...