Hi, I have a strange problem using FreeSwitch on an Amazon EC2 instance and was wondering if anyone out there could help me. The problem seems to be that I can register an extension from a remote network in say France, phone works ok but when passing calls to it (from sip peer through JavaScript to user extension) I get one way audio, the caller can hear me on the softphone but I can’t hear them – like the reverse audio path can’t be established. That would be ok and fixable if the issue was consistent but on another phone registered from within the UK, it all works fine and I get a good 2 way conversation. If I dial from one softphone in France directly to the other in the UK, that works too. I followed the wiki’s guide to installing on Amazon and have everything right, otherwise I doubt it would work at all. Is anyone able to help me or have experience of this type of issue? Maybe it’s a local router issue to the phone in France but I didn’t need to do anything special to the one in the UK to work, so not sure it could be that either. Kind Regards, Stuart
One way audio is generally a NAT/Firewall issue at the client end, and probably not related to EC2 at all. There has been a lot of work in the wiki recently regarding NAT issues. On 28 January 2013 09:26, Stuart Mills <stuart.mills3@btopenworld.com>wrote:
Hi,
I have a strange problem using FreeSwitch on an Amazon EC2 instance and was wondering if anyone out there could help me.
The problem seems to be that I can register an extension from a remote network in say France, phone works ok but when passing calls to it (from sip peer through JavaScript to user extension) I get one way audio, the caller can hear me on the softphone but I can’t hear them – like the reverse audio path can’t be established.
That would be ok and fixable if the issue was consistent but on another phone registered from within the UK, it all works fine and I get a good 2 way conversation.
If I dial from one softphone in France directly to the other in the UK, that works too.
I followed the wiki’s guide to installing on Amazon and have everything right, otherwise I doubt it would work at all.
Is anyone able to help me or have experience of this type of issue? Maybe it’s a local router issue to the phone in France but I didn’t need to do anything special to the one in the UK to work, so not sure it could be that either.
Kind Regards,
Stuart
_________________________________________________________________________ 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
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
-- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info@cassidywebservices.co.uk> *03300 100 960 *F<info@cassidywebservices.co.uk> *03300 100 961 *E <info@cassidywebservices.co.uk> *andrew@cassidywebservices.co.uk *W <info@cassidywebservices.co.uk> *www.cassidywebservices.co.uk
One way audio usually is a result of NAT. Not all NAT routers operate in the same way, and not all softphones handle it in the same way, so that may explain why it works in one location but not the other. I assume the softphones were behind NAT? Set the FS log level to debug, and check the IP in the remote SDP. You need the phone's external IP in there. Phones usually determine that using STUN. Some NAT routers have a feature called SIP ALG to rewrite it for phones not doing it themselves, but that often will conflict with phones that have already put the correct IP (and also cannot work with TLS). In short check where RTP is being sent to is correct... Without the correct IP FS also has a last-resort feature where it detects the IP the caller sends audio from and replies to there (RTP auto-adjust). A log entry indicates when that happens. But that requires FS to receive audio, which means it can't work until the caller starts speaking after the call is answered (so might miss hearing the callee say 'hello'), and the caller will never hear ringing (unless generated locally by the phone when it gets the SIP 183/180 response). The NAT page on the Wiki may be informative: http://wiki.freeswitch.org/wiki/NAT Just to complicate things Amazon EC2 runs behind a NAT layer too AFAIK. I don't have any personal experience with EC2. -Steve On 28 January 2013 09:26, Stuart Mills <stuart.mills3@btopenworld.com>wrote:
Hi,
I have a strange problem using FreeSwitch on an Amazon EC2 instance and was wondering if anyone out there could help me.
The problem seems to be that I can register an extension from a remote network in say France, phone works ok but when passing calls to it (from sip peer through JavaScript to user extension) I get one way audio, the caller can hear me on the softphone but I can’t hear them – like the reverse audio path can’t be established.
That would be ok and fixable if the issue was consistent but on another phone registered from within the UK, it all works fine and I get a good 2 way conversation.
If I dial from one softphone in France directly to the other in the UK, that works too.
I followed the wiki’s guide to installing on Amazon and have everything right, otherwise I doubt it would work at all.
Is anyone able to help me or have experience of this type of issue? Maybe it’s a local router issue to the phone in France but I didn’t need to do anything special to the one in the UK to work, so not sure it could be that either.
Kind Regards,
Stuart
_________________________________________________________________________ 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
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
Cheers Steven, I guessed it’s something to do with NAT – most likely at the softphone end. I’m using X-Lite in all softphone cases, so it’s literally the end user’s PC and router that’s different. I tried STUN but that seemed to make things worse where no audio was heard from both ends. I’ll run some debug and check the remote SDP to see what it contains and go from there. Stuart From: Steven Ayre Sent: Monday, January 28, 2013 10:12 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] softphones on Amazon One way audio usually is a result of NAT. Not all NAT routers operate in the same way, and not all softphones handle it in the same way, so that may explain why it works in one location but not the other. I assume the softphones were behind NAT? Set the FS log level to debug, and check the IP in the remote SDP. You need the phone's external IP in there. Phones usually determine that using STUN. Some NAT routers have a feature called SIP ALG to rewrite it for phones not doing it themselves, but that often will conflict with phones that have already put the correct IP (and also cannot work with TLS). In short check where RTP is being sent to is correct... Without the correct IP FS also has a last-resort feature where it detects the IP the caller sends audio from and replies to there (RTP auto-adjust). A log entry indicates when that happens. But that requires FS to receive audio, which means it can't work until the caller starts speaking after the call is answered (so might miss hearing the callee say 'hello'), and the caller will never hear ringing (unless generated locally by the phone when it gets the SIP 183/180 response). The NAT page on the Wiki may be informative: http://wiki.freeswitch.org/wiki/NAT Just to complicate things Amazon EC2 runs behind a NAT layer too AFAIK. I don't have any personal experience with EC2. -Steve On 28 January 2013 09:26, Stuart Mills <stuart.mills3@btopenworld.com> wrote: Hi, I have a strange problem using FreeSwitch on an Amazon EC2 instance and was wondering if anyone out there could help me. The problem seems to be that I can register an extension from a remote network in say France, phone works ok but when passing calls to it (from sip peer through JavaScript to user extension) I get one way audio, the caller can hear me on the softphone but I can’t hear them – like the reverse audio path can’t be established. That would be ok and fixable if the issue was consistent but on another phone registered from within the UK, it all works fine and I get a good 2 way conversation. If I dial from one softphone in France directly to the other in the UK, that works too. I followed the wiki’s guide to installing on Amazon and have everything right, otherwise I doubt it would work at all. Is anyone able to help me or have experience of this type of issue? Maybe it’s a local router issue to the phone in France but I didn’t need to do anything special to the one in the UK to work, so not sure it could be that either. Kind Regards, Stuart _________________________________________________________________________ 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 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 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
participants (3)
-
Andrew Cassidy
-
Steven Ayre
-
Stuart Mills