Hi all, I have a problem with RTP and NAT. The scenario is Hosted PBX and Natted phones (yealink): Phones (192.168.0.x) - NAT -> FS (public IP) When I call FS (for example an IVR) from a Phone, FS send the RTP to the private address (192.168.0.x) instead to the public NAT IP. The registration is ok: freeswitch@internal> sofia status profile tenant1.bs.dev.voip.clio.it reg 1 Registrations: ================================================================================================= Call-ID: 488014850@192.168.0.100 User: 202@tenant1.test.com Contact: "Test 202" <sip:202@192.168.0.100:5062;fs_nat=yes;fs_path=sip%3A202%40<NAT_IP>%3A37710> Agent: Yealink SIP-T20P 9.61.0.70 Status: Registered(UDP-NAT)(unknown) EXP(2012-06-06 19:01:55) EXPSECS(3232) Host: localhost.localdomain IP: <NAT_IP> Port: 37710 Auth-User: 202 Auth-Realm: tenant1.test.com MWI-Account: 202@tenant1.test.com How I can tell FS to send the RTP Packets to the right address? I think is needed a "comedia mode" like in Asterisk (or RTPProxy in openser)... Best regards,
The issue is more likely the phone, as the phone is responsible for handing FS the correct IP. There is however a way to force this on the FS side but may break other devices. Please take a look at NDLB (No Device Left Behind) parameters for Sofia on the wiki. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <jaasmailing@gmail.com> wrote:
Hi all,
I have a problem with RTP and NAT. The scenario is Hosted PBX and Natted phones (yealink):
Phones (192.168.0.x) - NAT -> FS (public IP)
When I call FS (for example an IVR) from a Phone, FS send the RTP to the private address (192.168.0.x) instead to the public NAT IP. The registration is ok:
freeswitch@internal> sofia status profile tenant1.bs.dev.voip.clio.it reg 1 Registrations:
================================================================================================= Call-ID: 488014850@192.168.0.100 User: 202@tenant1.test.com Contact: "Test 202" < sip:202@192.168.0.100:5062;fs_nat=yes;fs_path=sip%3A202%40 <NAT_IP>%3A37710> Agent: Yealink SIP-T20P 9.61.0.70 Status: Registered(UDP-NAT)(unknown) EXP(2012-06-06 19:01:55) EXPSECS(3232) Host: localhost.localdomain IP: <NAT_IP> Port: 37710 Auth-User: 202 Auth-Realm: tenant1.test.com MWI-Account: 202@tenant1.test.com
How I can tell FS to send the RTP Packets to the right address? I think is needed a "comedia mode" like in Asterisk (or RTPProxy in openser)...
Best regards,
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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NDLB-connectile-disfunction is probably what you're looking for, see: http://wiki.freeswitch.org/wiki/NDLB Brian Foster Endigo Computer LLC Sent from a mobile device. On Jun 6, 2012 12:28 PM, "Brian Foster" <bdfoster@endigotech.com> wrote:
The issue is more likely the phone, as the phone is responsible for handing FS the correct IP. There is however a way to force this on the FS side but may break other devices. Please take a look at NDLB (No Device Left Behind) parameters for Sofia on the wiki.
Brian Foster Endigo Computer LLC
Sent from a mobile device. On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <jaasmailing@gmail.com> wrote:
Hi all,
I have a problem with RTP and NAT. The scenario is Hosted PBX and Natted phones (yealink):
Phones (192.168.0.x) - NAT -> FS (public IP)
When I call FS (for example an IVR) from a Phone, FS send the RTP to the private address (192.168.0.x) instead to the public NAT IP. The registration is ok:
freeswitch@internal> sofia status profile tenant1.bs.dev.voip.clio.it reg 1 Registrations:
================================================================================================= Call-ID: 488014850@192.168.0.100 User: 202@tenant1.test.com Contact: "Test 202" < sip:202@192.168.0.100:5062;fs_nat=yes;fs_path=sip%3A202%40 <NAT_IP>%3A37710> Agent: Yealink SIP-T20P 9.61.0.70 Status: Registered(UDP-NAT)(unknown) EXP(2012-06-06 19:01:55) EXPSECS(3232) Host: localhost.localdomain IP: <NAT_IP> Port: 37710 Auth-User: 202 Auth-Realm: tenant1.test.com MWI-Account: 202@tenant1.test.com
How I can tell FS to send the RTP Packets to the right address? I think is needed a "comedia mode" like in Asterisk (or RTPProxy in openser)...
Best regards,
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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Ok, I'll try the parameter NDLB. Is it suitable in an hosted environment (thousand of extensions)? Anyway, the problem is that I can't use STUN (phone side configuration) because I have two needs: 1) Phone A (192.168.0.100) calls Phone B (192.168.0.101), in this case the RTP flow should be sent directly between two endpoints. If I use STUN the SDP "A" contains the public IP and not the private IP (that must be used for device reachability). 2) Phone A (192.168.0.100) calls Freeswitch (Public IP), in this case the RTP flow shoud be sent to Freeswitch Public IP and Freeswitch shoud send the RTP to the Phone A NAT IP. Here the STUN is the best solution. What is the best practice in this scenario? Regards, Il 06/06/12 18.28, Brian Foster ha scritto:
The issue is more likely the phone, as the phone is responsible for handing FS the correct IP. There is however a way to force this on the FS side but may break other devices. Please take a look at NDLB (No Device Left Behind) parameters for Sofia on the wiki.
Brian Foster Endigo Computer LLC
Sent from a mobile device.
On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <jaasmailing@gmail.com <mailto:jaasmailing@gmail.com>> wrote:
Hi all,
I have a problem with RTP and NAT. The scenario is Hosted PBX and Natted phones (yealink):
Phones (192.168.0.x) - NAT -> FS (public IP)
When I call FS (for example an IVR) from a Phone, FS send the RTP to the private address (192.168.0.x) instead to the public NAT IP. The registration is ok:
For a hosted environment, where you're not in control of the users devices/routers you should do the following: In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true. That should work every time. --Dave _____ From: Carlo Dimaggio [mailto:jaasmailing@gmail.com] To: freeswitch-users@lists.freeswitch.org Sent: Wed, 06 Jun 2012 13:13:08 -0700 Subject: Re: [Freeswitch-users] RTP NAT issue Ok, I'll try the parameter NDLB. Is it suitable in an hosted environment (thousand of extensions)? Anyway, the problem is that I can't use STUN (phone side configuration) because I have two needs: 1) Phone A (192.168.0.100) calls Phone B (192.168.0.101), in this case the RTP flow should be sent directly between two endpoints. If I use STUN the SDP "A" contains the public IP and not the private IP (that must be used for device reachability). 2) Phone A (192.168.0.100) calls Freeswitch (Public IP), in this case the RTP flow shoud be sent to Freeswitch Public IP and Freeswitch shoud send the RTP to the Phone A NAT IP. Here the STUN is the best solution. What is the best practice in this scenario? Regards, Il 06/06/12 18.28, Brian Foster ha scritto: The issue is more likely the phone, as the phone is responsible for handing FS the correct IP. There is however a way to force this on the FS side but may break other devices. Please take a look at NDLB (No Device Left Behind) parameters for Sofia on the wiki. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <jaasmailing@gmail.com> wrote: Hi all, I have a problem with RTP and NAT. The scenario is Hosted PBX and Natted phones (yealink): Phones (192.168.0.x) - NAT -> FS (public IP) When I call FS (for example an IVR) from a Phone, FS send the RTP to the private address (192.168.0.x) instead to the public NAT IP. The registration is ok:
Suboptimal solution. you'll break transfers among other things. Sent from my iPad On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel" <drk@drkngs.net> wrote:
For a hosted environment, where you're not in control of the users devices/routers you should do the following:
In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true.
That should work every time.
--Dave
Maybe doing a sip proxy on the nat'd side? Brian Foster Endigo Computer LLC Sent from a mobile device. On Jun 6, 2012 9:57 PM, "Brian West" <brian@freeswitch.org> wrote:
Suboptimal solution. you'll break transfers among other things.
Sent from my iPad
On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel" <drk@drkngs.net> wrote:
For a hosted environment, where you're not in control of the users devices/routers you should do the following:
In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true.
That should work every time.
--Dave
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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Been using this configuration for service providers for years, and always works. Everything always works, transfers etc... --Dave _____ From: Brian West [mailto:brian@freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] Sent: Wed, 06 Jun 2012 18:55:41 -0700 Subject: Re: [Freeswitch-users] RTP NAT issue Suboptimal solution. you'll break transfers among other things. Sent from my iPad On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel" <drk@drkngs.net> wrote:
For a hosted environment, where you're not in control of the users devices/routers you should do the following:
In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true.
That should work every time.
--Dave
_________________________________________________________________________ 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 Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Dave, I've tried the "aggressive_nat_detection" on my internal profile. Restarted the profile and I have this registration: Contact: "Test 202" <sip:202@192.168.2.100:5062;fs_nat=yes;fs_path=sip%3A202%40<FS IP>%3A37710> but FS send the RTP to the private IP (192.168.2.100) and not to the public NAT IP. Is there something wrong? Best Regards Il 07/06/12 05.23, Dave R. Kompel ha scritto:
Been using this configuration for service providers for years, and always works. Everything always works, transfers etc... --Dave
------------------------------------------------------------------------ *From:* Brian West [mailto:brian@freeswitch.org] *To:* FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] *Sent:* Wed, 06 Jun 2012 18:55:41 -0700 *Subject:* Re: [Freeswitch-users] RTP NAT issue
Suboptimal solution. you'll break transfers among other things.
Sent from my iPad
On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel" <drk@drkngs.net <mailto:drk@drkngs.net>> wrote:
> For a hosted environment, where you're not in control of the users devices/routers you should do the following: > > In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true. > > That should work every time. > > > --Dave
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
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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
Join Us At ClueCon - Aug 7-9, 2012
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Well FreeSWITCH needs to get an RTP packet to know which port/IP to respond to, for the general case. Do a full UDP capture and open it in Wireshark. The VoIP calls flow view will show you RTP streams, too. -Michael From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Carlo Dimaggio Sent: Thursday, June 07, 2012 1:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP NAT issue Hi Dave, I've tried the "aggressive_nat_detection" on my internal profile. Restarted the profile and I have this registration: Contact: "Test 202" <sip:202@192.168.2.100:5062;fs_nat=yes;fs_path=sip%3A202%40<FS IP>%3A37710> but FS send the RTP to the private IP (192.168.2.100) and not to the public NAT IP. Is there something wrong? Best Regards Il 07/06/12 05.23, Dave R. Kompel ha scritto: Been using this configuration for service providers for years, and always works. Everything always works, transfers etc... --Dave ________________________________ From: Brian West [mailto:brian@freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] Sent: Wed, 06 Jun 2012 18:55:41 -0700 Subject: Re: [Freeswitch-users] RTP NAT issue Suboptimal solution. you'll break transfers among other things. Sent from my iPad On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel" <drk@drkngs.net<mailto:drk@drkngs.net>> wrote:
For a hosted environment, where you're not in control of the users devices/routers you should do the following:
In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true.
That should work every time.
--Dave
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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 Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Brian, what is for you the optimal solution in an hosted environment? Regards Il 07/06/12 03.55, Brian West ha scritto:
Suboptimal solution. you'll break transfers among other things.
Sent from my iPad
On Jun 6, 2012, at 4:01 PM, "Dave R. Kompel"<drk@drkngs.net> wrote:
For a hosted environment, where you're not in control of the users devices/routers you should do the following:
In the SIP profile, turn on agressive_nat_detection, on the client device have them turn off ALL nat mapping stuff, so the switch can detect it's nat, and if it don't work cause they have some broken router that's doing ALG, and only in that case then set the directory entry for that use to have the "NDLB-Connnectile-Dysfunction" be true.
That should work every time.
--Dave
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
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Try this: - Enable agressive_nat_detection on the profile as per Dave's suggestion - Make sure SIP ALG is not enabled on the NAT router - On the Yealink set Rport enabled (Click Accounts, Advanced, set Rport drop down to enabled) I find this works 99% of the time. When I look at my registrations from a Yealink the contact line is like this: "John" <sip:1001@192.168.15.6:5062;received=PUBLIC_IP:1032;fs_nat=yes;fs_path=sip%3A1001%40 PUBLIC_IP%3A1032> Ash. On 07/06/2012, at 2:28 AM, Brian Foster wrote:
The issue is more likely the phone, as the phone is responsible for handing FS the correct IP. There is however a way to force this on the FS side but may break other devices. Please take a look at NDLB (No Device Left Behind) parameters for Sofia on the wiki.
Brian Foster Endigo Computer LLC
Sent from a mobile device.
On Jun 6, 2012 12:15 PM, "Carlo Dimaggio" <jaasmailing@gmail.com> wrote: Hi all,
I have a problem with RTP and NAT. The scenario is Hosted PBX and Natted phones (yealink):
Phones (192.168.0.x) - NAT -> FS (public IP)
When I call FS (for example an IVR) from a Phone, FS send the RTP to the private address (192.168.0.x) instead to the public NAT IP. The registration is ok:
freeswitch@internal> sofia status profile tenant1.bs.dev.voip.clio.it reg 1 Registrations: ================================================================================================= Call-ID: 488014850@192.168.0.100 User: 202@tenant1.test.com Contact: "Test 202" <sip:202@192.168.0.100:5062;fs_nat=yes;fs_path=sip%3A202%40<NAT_IP>%3A37710> Agent: Yealink SIP-T20P 9.61.0.70 Status: Registered(UDP-NAT)(unknown) EXP(2012-06-06 19:01:55) EXPSECS(3232) Host: localhost.localdomain IP: <NAT_IP> Port: 37710 Auth-User: 202 Auth-Realm: tenant1.test.com MWI-Account: 202@tenant1.test.com
How I can tell FS to send the RTP Packets to the right address? I think is needed a "comedia mode" like in Asterisk (or RTPProxy in openser)...
Best regards,
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Regards, Ashley Breeden URL Networks Pty Ltd Email: ash@url.net.au WWW: http://www.url.net.au Phone: 1300 33 11 78 Direct: 03 9008 5901 Mobile: 0411 112 056 HostedPBX - VoIP - Hosting - Wholesale
participants (6)
-
Ashley Breeden
-
Brian Foster
-
Brian West
-
Carlo Dimaggio
-
Dave R. Kompel
-
Michael Giagnocavo