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[Freeswitch-users] Freeswitch behind NAT with 2 external providers: No audio


 
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PostPosted: Wed Mar 09, 2016 9:33 am    Post subject: [Freeswitch-users] Freeswitch behind NAT with 2 external pro Reply with quote

Hello,

I have the following scenario
  • Freeswitch is behind NAT with external profile with ext_sip_ip and ext_rtp_ip
  • Echo from an external SIP provider test inbound works with audio
  • Calling outbound to the same external SIP provider works with audio
  • Receiving a call from the external SIP provider and bridging it to another number at the same external SIP provider does not have audio
  • Receiving a call from the external SIP provider and bridging it to a local Patton PRI works with audio

I also set up another (modified NAT) profile
  • like external profile - but with a different ext_rtp_ip
  • calling outbound to via NAT profile to another SIP provider works with audio
  • Receiving a call from the external SIP provider (external profile) and bridging it to another number at the another external SIP provider [/b](nat profile) does not have audio[/b]
  • Tcpdump btw. does not show an attempt of Freeswitch to issue any outbound RTP packet. I can see only SIP messages towards the outbound SIP provider

So I think, there is a problem with briding a natted external call to another natted external call, both with ext-rtp-ip. It tried this on an older Freeswitch (1 year old) and with a more recent Freeswitch (1 month old), both behave the same. Same scenario with a non-natted Freeswitch works fine.
Btw. Freeswitch logs do show
RTCPpacket not written
right after bridginbg the call.

Question: Does anybody have a clue, what could cause the problem? Maybe there is any parameter in the profiles, that may help?



Quote:
--
With kind regards
Peter Steinbach

Telefaks Services GmbH
mailto:lists ([email]lists[/email]) (att) telefaks.de
Internet: www.telefaks.de

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brian at freeswitch.org
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PostPosted: Wed Mar 09, 2016 9:48 am    Post subject: [Freeswitch-users] Freeswitch behind NAT with 2 external pro Reply with quote

Its going to be because you probably need to pre_answer, play a little media before you hair pin the call back out the same provider. Smile  playback silence_stream://1000 after pre_answer would probably fix it prior to bridge.

On Wed, Mar 9, 2016 at 8:32 AM, Peter Steinbach <lists@telefaks.de (lists@telefaks.de)> wrote:
Quote:
Hello,

I have the following scenario
  • Freeswitch is behind NAT with external profile with ext_sip_ip and ext_rtp_ip
  • Echo from an external SIP provider test inbound works with audio
  • Calling outbound to the same external SIP provider works with audio
  • Receiving a call from the external SIP provider and bridging it to another number at the same external SIP provider does not have audio
  • Receiving a call from the external SIP provider and bridging it to a local Patton PRI works with audio

I also set up another (modified NAT) profile
  • like external profile - but with a different ext_rtp_ip
  • calling outbound to via NAT profile to another SIP provider works with audio
  • Receiving a call from the external SIP provider (external profile) and bridging it to another number at the another external SIP provider [/b](nat profile) does not have audio[/b]
  • Tcpdump btw. does not show an attempt of Freeswitch to issue any outbound RTP packet. I can see only SIP messages towards the outbound SIP provider

So I think, there is a problem with briding a natted external call  to another natted external call, both with ext-rtp-ip. It tried this on an older Freeswitch (1 year old) and with a more recent Freeswitch (1 month old), both behave the same. Same scenario with a non-natted Freeswitch works fine.
Btw. Freeswitch logs do show
    RTCPpacket not written
right after bridginbg the call.

Question: Does anybody have a clue, what could cause the problem? Maybe there is any parameter in the profiles, that may help?



Quote:
--
With kind regards
Peter Steinbach

Telefaks Services GmbH
mailto:lists ([email]lists[/email]) (att) telefaks.de
Internet: www.telefaks.de



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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




--

Brian West
brian@freeswitch.org (brian@freeswitch.org)


Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here! | Reddit: /r/freeswitch
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
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lists at telefaks.de
Guest





PostPosted: Wed Mar 09, 2016 11:37 am    Post subject: [Freeswitch-users] Freeswitch behind NAT with 2 external pro Reply with quote

Herr Brian,

thank you, that did the trick!
I added a pre_aswer, a playback and finally a ring_ready in order to playback a ring tone to the caller. Now we have 2 way audio.

Best regards
Peter


On 03/09/16 15:47, Brian West wrote:

Quote:
Its going to be because you probably need to pre_answer, play a little media before you hair pin the call back out the same provider. Smile playback silence_stream://1000 after pre_answer would probably fix it prior to bridge.

On Wed, Mar 9, 2016 at 8:32 AM, Peter Steinbach <lists@telefaks.de (lists@telefaks.de)> wrote:
Quote:
Hello,

I have the following scenario
  • Freeswitch is behind NAT with external profile with ext_sip_ip and ext_rtp_ip
  • Echo from an external SIP provider test inbound works with audio
  • Calling outbound to the same external SIP provider works with audio
  • Receiving a call from the external SIP provider and bridging it to another number at the same external SIP provider does not have audio
  • Receiving a call from the external SIP provider and bridging it to a local Patton PRI works with audio

I also set up another (modified NAT) profile
  • like external profile - but with a different ext_rtp_ip
  • calling outbound to via NAT profile to another SIP provider works with audio
  • Receiving a call from the external SIP provider (external profile) and bridging it to another number at the another external SIP provider (nat profile) does not have audio[/b]
  • Tcpdump btw. does not show an attempt of Freeswitch to issue any outbound RTP packet. I can see only SIP messages towards the outbound SIP provider

So I think, there is a problem with briding a natted external call to another natted external call, both with ext-rtp-ip. It tried this on an older Freeswitch (1 year old) and with a more recent Freeswitch (1 month old), both behave the same. Same scenario with a non-natted Freeswitch works fine.
Btw. Freeswitch logs do show
RTCPpacket not written
right after bridginbg the call.

Question: Does anybody have a clue, what could cause the problem? Maybe there is any parameter in the profiles, that may help?



Quote:
--
With kind regards
Peter Steinbach

Telefaks Services GmbH
mailto:lists ([email]lists[/email]) (att) telefaks.de
Internet: www.telefaks.de



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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




--

Brian West
brian@freeswitch.org (brian@freeswitch.org)


Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here! | Reddit: /r/freeswitch
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest














Quote:
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (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
--
With kind regards
Peter Steinbach

Telefaks Services GmbH
mailto:lists ([email]lists[/email]) (att) telefaks.de
Internet: www.telefaks.de

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