No audio/dtmf from softphone behind NAT
I've got a FreeSwitch server up on Amazon EC2, ports wide open for my office external-IP, server iptables disabled, and changed the FreeSwitch ACL domains to "allow", so it's all wide open for now. In the office I'm trying to connect to the server from Bria/X-lite. I've entered a stun server (stun.freeswitch.org) and I can now call to the server, but not from the server. I read this page: http://wiki.freeswitch.org/wiki/Nat_stun_debug_irc which suggested adding this variable to the user config: http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA#NDLB-connectile-dysfunc... With that on I can now call to and from the server. However with or without that although I can hear audio from the server, audio to the server isn't arriving (doesn't appear in recordings), and dtmf doesn't get received either. When I hang up from the client, I see in the CLI that it gets that instruction, so it hasn't started the call and lost all contact with the softphone, it's receiving some instructions, but not the audio and dtmf. The problem is that both the server and client are each behind NAT, so either could be having the problem (on EC2 the auto-NAT doesn't work, so I've specified the external rtp and sip ip's.. I've also turned on aggressive-NAT in case that helps. Also I'm connecting to the server by a sub-domain (A-name) rather than IP.) I've got almost the same setup working fine on the internal network (same dialplan and directory, and all the config is the same if it can be), so it's got to be something to do with the NAT's. Any suggestions on what the problem might be, or how to find it? Cheers, Fraser
I've taken another stab at this one way audio problem today. I've run a wireshark capture, and looking at the RTP analysis it only has the down-stream, it doesn't record anything being sent upstream at all. Below is the SIP graph, which shows RTP coming down, but none going up. But I don't know enough about SIP to know whether something is missing. Any suggestions of what I should try now? Would the dtmf's be sent in the sip packets, or in the rtp? To preempt the easy answers and save some time: - ports are open on EC2 config, - iptables turned off for the test, - RTP port range uncommented in switch.conf.xml, - softphone stun was set to stun.freeswitch.org Cheers, Fraser |Time | 192.168.1.8 | | | | 184.73.226.197 | |6.488 | INVITE SDP ( BV32 BV32-FEC SPEEX SPEEX-FEC g71...iLBC g711A g) |SIP From: sip:12610@184.73.226.197<sip%3A12610@184.73.226.197> To:sip:12605@184.73.226.197 <To%3Asip%3A12605@184.73.226.197> | |(25829) ------------------> (5060) | |6.615 | 100 Trying| |SIP Status | |(25829) <------------------ (5060) | |6.623 | 407 Proxy Authentication Required |SIP Status | |(25829) <------------------ (5060) | |6.623 | ACK | |SIP Request | |(25829) ------------------> (5060) | |6.738 | INVITE SDP ( BV32 BV32-FEC SPEEX SPEEX-FEC g71...iLBC g711A g) |SIP From: sip:12610@184.73.226.197<sip%3A12610@184.73.226.197> To:sip:12605@184.73.226.197 <To%3Asip%3A12605@184.73.226.197> | |(25829) ------------------> (5060) | |6.869 | 100 Trying| |SIP Status | |(25829) <------------------ (5060) | |7.070 | 183 Session Progress SDP ( g711U telephone-eve... |SIP Status | |(25829) <------------------ (5060) | |7.264 | RTP (g711U) |RTP Num packets:520 Duration:10.793s SSRC:0x5433093E | |(44172) <------------------ (30432) | |18.090 | 200 OK SDP ( g711U telephone-event) |SIP Status | |(25829) <------------------ (5060) | |18.112 | ACK | |SIP Request | |(25829) ------------------> (5060) | |31.750 | BYE | |SIP Request | |(25829) ------------------> (5060) | |31.872 | 200 OK | |SIP Status | |(25829) <------------------ (5060) | On Sat, Apr 3, 2010 at 4:49 PM, Fraser Redmond <fraserredmond@gmail.com>wrote:
I've got a FreeSwitch server up on Amazon EC2, ports wide open for my office external-IP, server iptables disabled, and changed the FreeSwitch ACL domains to "allow", so it's all wide open for now.
In the office I'm trying to connect to the server from Bria/X-lite. I've entered a stun server (stun.freeswitch.org) and I can now call to the server, but not from the server. I read this page: http://wiki.freeswitch.org/wiki/Nat_stun_debug_irc which suggested adding this variable to the user config:
http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA#NDLB-connectile-dysfunc...
With that on I can now call to and from the server. However with or without that although I can hear audio from the server, audio to the server isn't arriving (doesn't appear in recordings), and dtmf doesn't get received either.
When I hang up from the client, I see in the CLI that it gets that instruction, so it hasn't started the call and lost all contact with the softphone, it's receiving some instructions, but not the audio and dtmf.
The problem is that both the server and client are each behind NAT, so either could be having the problem (on EC2 the auto-NAT doesn't work, so I've specified the external rtp and sip ip's.. I've also turned on aggressive-NAT in case that helps. Also I'm connecting to the server by a sub-domain (A-name) rather than IP.)
I've got almost the same setup working fine on the internal network (same dialplan and directory, and all the config is the same if it can be), so it's got to be something to do with the NAT's.
Any suggestions on what the problem might be, or how to find it?
Cheers, Fraser
Most likely its in the RTP stream as RFC2833 which is the reason you're not getting anything plus I need a FULL sip trace not this abbreviated trace. /b On Apr 5, 2010, at 11:12 AM, Fraser Redmond wrote:
I've taken another stab at this one way audio problem today.
I've run a wireshark capture, and looking at the RTP analysis it only has the down-stream, it doesn't record anything being sent upstream at all.
Below is the SIP graph, which shows RTP coming down, but none going up. But I don't know enough about SIP to know whether something is missing.
Any suggestions of what I should try now?
Would the dtmf's be sent in the sip packets, or in the rtp?
To preempt the easy answers and save some time: - ports are open on EC2 config, - iptables turned off for the test, - RTP port range uncommented in switch.conf.xml, - softphone stun was set to stun.freeswitch.org
Cheers, Fraser
Thanks Brian. Sorry, should have done a full sip trace before, but here is one now: <http://pastebin.freeswitch.org/12632> Calling an IVR dialplan: http://pastebin.freeswitch.org/12634 Calling from one extn to another. http://pastebin.freeswitch.org/12633 (With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.) For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all. Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.) Cheers, Fraser On Mon, Apr 5, 2010 at 5:21 PM, Brian West <brian@freeswitch.org> wrote:
Most likely its in the RTP stream as RFC2833 which is the reason you're not getting anything plus I need a FULL sip trace not this abbreviated trace.
/b
On Apr 5, 2010, at 11:12 AM, Fraser Redmond wrote:
I've taken another stab at this one way audio problem today.
I've run a wireshark capture, and looking at the RTP analysis it only has the down-stream, it doesn't record anything being sent upstream at all.
Below is the SIP graph, which shows RTP coming down, but none going up. But I don't know enough about SIP to know whether something is missing.
Any suggestions of what I should try now?
Would the dtmf's be sent in the sip packets, or in the rtp?
To preempt the easy answers and save some time: - ports are open on EC2 config, - iptables turned off for the test, - RTP port range uncommented in switch.conf.xml, - softphone stun was set to stun.freeswitch.org
Cheers, Fraser
_______________________________________________ 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
sofia loglevel all 0 sofia profile xx siptrace on replace xx with profile. What you have provided is NOT a sip trace. Thanks, Brian On Apr 5, 2010, at 12:28 PM, Fraser Redmond wrote:
Thanks Brian. Sorry, should have done a full sip trace before, but here is one now:
Calling an IVR dialplan: http://pastebin.freeswitch.org/12634
Calling from one extn to another. http://pastebin.freeswitch.org/12633 (With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.)
For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all.
Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.)
Cheers, Fraser
Ok, sorry, try this: http://pastebin.freeswitch.org/12635 And, yes, as I said in my earlier message, the firewall is off - checked it over and over again :-) Amazon EC2 has it's own firewall system as well, which can't be turned off, but I've set it to enable all ports (0-65535) Cheers, Fraser On Mon, Apr 5, 2010 at 6:34 PM, Brian West <brian@freeswitch.org> wrote:
sofia loglevel all 0 sofia profile xx siptrace on
replace xx with profile. What you have provided is NOT a sip trace.
Thanks, Brian
On Apr 5, 2010, at 12:28 PM, Fraser Redmond wrote:
Thanks Brian. Sorry, should have done a full sip trace before, but here is one now: <http://pastebin.freeswitch.org/12632>
Calling an IVR dialplan: http://pastebin.freeswitch.org/12634
Calling from one extn to another. http://pastebin.freeswitch.org/12633 (With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.)
For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all.
Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.)
Cheers, Fraser
_______________________________________________ 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
While its valid to have host names in the SDP we do NOT and WILL NOT support it please input IP addresses into the rtp-ip and ext-rtp-ip fields. You'll find that it WORKS BETTER. /b On Apr 5, 2010, at 12:53 PM, Fraser Redmond wrote:
Ok, sorry, try this: http://pastebin.freeswitch.org/12635
And, yes, as I said in my earlier message, the firewall is off - checked it over and over again :-)
Amazon EC2 has it's own firewall system as well, which can't be turned off, but I've set it to enable all ports (0-65535)
Cheers, Fraser
Thanks Brian, it's annoying that there's some places that say to only have IP's and elsewhere it says domain is ok. I'll change it over and try again now, but I had tried putting those back to IP's (I've tried everything!) in some of my earlier testing the other day it hadn't helped. But I'll double-check now. Do you want another trace using IP's if it's still not working? Cheers, Fraser On Mon, Apr 5, 2010 at 7:08 PM, Brian West <brian@freeswitch.org> wrote:
While its valid to have host names in the SDP we do NOT and WILL NOT support it please input IP addresses into the rtp-ip and ext-rtp-ip fields. You'll find that it WORKS BETTER.
/b
On Apr 5, 2010, at 12:53 PM, Fraser Redmond wrote:
Ok, sorry, try this: http://pastebin.freeswitch.org/12635
And, yes, as I said in my earlier message, the firewall is off - checked it over and over again :-)
Amazon EC2 has it's own firewall system as well, which can't be turned off, but I've set it to enable all ports (0-65535)
Cheers, Fraser
_______________________________________________ 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
Plus you can't trust devices will resolve that... which is part of what I suspect is going on. /b On Apr 5, 2010, at 1:20 PM, Fraser Redmond wrote:
Thanks Brian, it's annoying that there's some places that say to only have IP's and elsewhere it says domain is ok.
I'll change it over and try again now, but I had tried putting those back to IP's (I've tried everything!) in some of my earlier testing the other day it hadn't helped. But I'll double-check now. Do you want another trace using IP's if it's still not working?
Cheers, Fraser
That works! Awesome! Thank you so much :-) Cheers, Fraser On Mon, Apr 5, 2010 at 7:20 PM, Fraser Redmond <fraserredmond@gmail.com>wrote:
Thanks Brian, it's annoying that there's some places that say to only have IP's and elsewhere it says domain is ok.
I'll change it over and try again now, but I had tried putting those back to IP's (I've tried everything!) in some of my earlier testing the other day it hadn't helped. But I'll double-check now. Do you want another trace using IP's if it's still not working?
Cheers, Fraser
On Mon, Apr 5, 2010 at 7:08 PM, Brian West <brian@freeswitch.org> wrote:
While its valid to have host names in the SDP we do NOT and WILL NOT support it please input IP addresses into the rtp-ip and ext-rtp-ip fields. You'll find that it WORKS BETTER.
/b
On Apr 5, 2010, at 12:53 PM, Fraser Redmond wrote:
Ok, sorry, try this: http://pastebin.freeswitch.org/12635
And, yes, as I said in my earlier message, the firewall is off - checked it over and over again :-)
Amazon EC2 has it's own firewall system as well, which can't be turned off, but I've set it to enable all ports (0-65535)
Cheers, Fraser
_______________________________________________ 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
Your very welcome. You might think about adding a wiki entry in the FAQ or the setup guide that outlines that. /b On Apr 5, 2010, at 1:31 PM, Fraser Redmond wrote:
That works! Awesome! Thank you so much :-)
Cheers, Fraser
On Mon, Apr 5, 2010 at 7:20 PM, Fraser Redmond <fraserredmond@gmail.com> wrote: Thanks Brian, it's annoying that there's some places that say to only have IP's and elsewhere it says domain is ok.
I'll change it over and try again now, but I had tried putting those back to IP's (I've tried everything!) in some of my earlier testing the other day it hadn't helped. But I'll double-check now. Do you want another trace using IP's if it's still not working?
Cheers, Fraser
Also have you checked your firewall? service iptables stop and test . /b On Apr 5, 2010, at 12:28 PM, Fraser Redmond wrote:
Thanks Brian. Sorry, should have done a full sip trace before, but here is one now:
Calling an IVR dialplan: http://pastebin.freeswitch.org/12634
Calling from one extn to another. http://pastebin.freeswitch.org/12633 (With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.)
For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all.
Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.)
Cheers, Fraser
participants (2)
-
Brian West
-
Fraser Redmond