FreeSwitch Proxy + RTPProxy Media server
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-Medi... Sent from the freeswitch-users mailing list archive at Nabble.com.
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 <adahary@gmail.com>
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
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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-Medi... Sent from the freeswitch-users mailing list archive at Nabble.com.
So why not just <action application="redirect" data="sip:foo@bar.com "/> 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 <adahary@gmail.com> 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
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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 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Medi... Sent from the freeswitch-users mailing list archive at Nabble.com.
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" <adahary@gmail.com> 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
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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-Medi... Sent from the freeswitch-users mailing list archive at Nabble.com.
You miss my point just use FreeSWITCH instead of RTPproxy... On 5/23/13 12:16 AM, "adahary" <adahary@gmail.com> 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
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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? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Medi... Sent from the freeswitch-users mailing list archive at Nabble.com.
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" <adahary@gmail.com> 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 endpointB@rtpproxy.fs.com' ? would that make just the RTP relay between endpoints without initiating a new SIP session with rtpproxy.fs.com?
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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) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Medi... Sent from the freeswitch-users mailing list archive at Nabble.com.
They may or may not be the same box. On 5/23/13 1:52 AM, "adahary" <adahary@gmail.com> 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 ENDPOINT.B@FS.B' ENDPOINT.A <-> FS B <-> ENDPOINT.B (rtp relay)
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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). -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Medi... Sent from the freeswitch-users mailing list archive at Nabble.com.
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] -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Medi... Sent from the freeswitch-users mailing list archive at Nabble.com.
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" <adahary@gmail.com> 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/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]
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Medi... Sent from the freeswitch-users mailing list archive at Nabble.com.
Assaf, Just curious if you are able to run your setup by having this scenario. I'm also looking for similar solution and wondering if we can achieve with FS without using any other media proxies. Please update this thread if possible. Thanks On Mon, Jun 3, 2013 at 4:51 AM, adahary <adahary@gmail.com> wrote:
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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org 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
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participants (5)
-
adahary
-
Dmitry Sytchev
-
jay binks
-
Jeff Ullman
-
Ken Rice