default muted media on conference ?!
Hello, I am trying to establish a conference group call using the setup pictured at [1]. Thus, when a user calls to a group extension, all the users in that group are called. I have set the sip-ip and rtp-ip bind IPs. Signaling works fine, the phones in the group ring. When I answer I even hear the wave playback conference starter/joiner. I can see bindings on the RTP public IP:PORT (using netstat). I can ngrep packets coming in and out for that ports. The problem I have is that I can hear no media. I've been debugging and eliminated the possible networking firewall issues using netcat over UDP and some high ports, between public and private RTP ip, in both directions (firewall allows everything). Thus, I would say that the problem is on the application side. This kind of looks like a scenario where all conference users are muted, even if I don't have any rules like that in the dialplan group extension. The freeswitch ACLs default="allow" and I don't see any "blocked by acl" logs. The logs look fine: "... muxing call legs". Any ideas what might be the problem? Thanks in advance, Stefan [1] https://postimg.org/image/4cwxn91bb/
What I think is the problem is the vpn. If you are using sip it is 99% the cause. If you can try without it just as a test. Also try 1on1 call and different uac if possible. Just to eliminate possibilities. Regards, Mirko On Jul 2, 2016 9:30 AM, "Anonim Stefan" <fanx07@gmail.com> wrote:
Hello,
I am trying to establish a conference group call using the setup pictured at [1]. Thus, when a user calls to a group extension, all the users in that group are called.
I have set the sip-ip and rtp-ip bind IPs. Signaling works fine, the phones in the group ring. When I answer I even hear the wave playback conference starter/joiner. I can see bindings on the RTP public IP:PORT (using netstat). I can ngrep packets coming in and out for that ports.
The problem I have is that I can hear no media.
I've been debugging and eliminated the possible networking firewall issues using netcat over UDP and some high ports, between public and private RTP ip, in both directions (firewall allows everything).
Thus, I would say that the problem is on the application side. This kind of looks like a scenario where all conference users are muted, even if I don't have any rules like that in the dialplan group extension. The freeswitch ACLs default="allow" and I don't see any "blocked by acl" logs. The logs look fine: "... muxing call legs".
Any ideas what might be the problem?
Thanks in advance, Stefan
[1] https://postimg.org/image/4cwxn91bb/
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Hi Mirko, I will see if I can try the things you proposed. Could you please tell me what makes you think it's the VPN and what you think it might be the misconfiguration in the VPN configuration (e.g. "small packets not allowed" etc...)? I see that SIP signaling works fine(and advertises the private RTP IP) and I can grep for RTP packets coming in and out FS machine, on the RTP public IP. P.S. In the same configuration, if instead of Freeswitch I have RPTEngine (as a media relay) it works fine. Thank you, Stefan On Sat, Jul 2, 2016 at 11:31 AM, Mirko Brankovic <mirkobrankovic@gmail.com> wrote:
What I think is the problem is the vpn. If you are using sip it is 99% the cause.
If you can try without it just as a test. Also try 1on1 call and different uac if possible. Just to eliminate possibilities.
Regards, Mirko On Jul 2, 2016 9:30 AM, "Anonim Stefan" <fanx07@gmail.com> wrote:
Hello,
I am trying to establish a conference group call using the setup pictured at [1]. Thus, when a user calls to a group extension, all the users in that group are called.
I have set the sip-ip and rtp-ip bind IPs. Signaling works fine, the phones in the group ring. When I answer I even hear the wave playback conference starter/joiner. I can see bindings on the RTP public IP:PORT (using netstat). I can ngrep packets coming in and out for that ports.
The problem I have is that I can hear no media.
I've been debugging and eliminated the possible networking firewall issues using netcat over UDP and some high ports, between public and private RTP ip, in both directions (firewall allows everything).
Thus, I would say that the problem is on the application side. This kind of looks like a scenario where all conference users are muted, even if I don't have any rules like that in the dialplan group extension. The freeswitch ACLs default="allow" and I don't see any "blocked by acl" logs. The logs look fine: "... muxing call legs".
Any ideas what might be the problem?
Thanks in advance, Stefan
[1] https://postimg.org/image/4cwxn91bb/
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Hi Stefan, Well, just the know issue that some nat configurations (traversal i think) may have problems with gathering ice candidates so even when your sip signaling is ok sdp might be carrying wrong candidates. But maybe this is not your case, if it works with different media server. Mirko On Jul 2, 2016 5:27 PM, "Anonim Stefan" <fanx07@gmail.com> wrote:
Hi Mirko,
I will see if I can try the things you proposed.
Could you please tell me what makes you think it's the VPN and what you think it might be the misconfiguration in the VPN configuration (e.g. "small packets not allowed" etc...)? I see that SIP signaling works fine(and advertises the private RTP IP) and I can grep for RTP packets coming in and out FS machine, on the RTP public IP.
P.S. In the same configuration, if instead of Freeswitch I have RPTEngine (as a media relay) it works fine.
Thank you, Stefan
On Sat, Jul 2, 2016 at 11:31 AM, Mirko Brankovic <mirkobrankovic@gmail.com
wrote:
What I think is the problem is the vpn. If you are using sip it is 99% the cause.
If you can try without it just as a test. Also try 1on1 call and different uac if possible. Just to eliminate possibilities.
Regards, Mirko On Jul 2, 2016 9:30 AM, "Anonim Stefan" <fanx07@gmail.com> wrote:
Hello,
I am trying to establish a conference group call using the setup pictured at [1]. Thus, when a user calls to a group extension, all the users in that group are called.
I have set the sip-ip and rtp-ip bind IPs. Signaling works fine, the phones in the group ring. When I answer I even hear the wave playback conference starter/joiner. I can see bindings on the RTP public IP:PORT (using netstat). I can ngrep packets coming in and out for that ports.
The problem I have is that I can hear no media.
I've been debugging and eliminated the possible networking firewall issues using netcat over UDP and some high ports, between public and private RTP ip, in both directions (firewall allows everything).
Thus, I would say that the problem is on the application side. This kind of looks like a scenario where all conference users are muted, even if I don't have any rules like that in the dialplan group extension. The freeswitch ACLs default="allow" and I don't see any "blocked by acl" logs. The logs look fine: "... muxing call legs".
Any ideas what might be the problem?
Thanks in advance, Stefan
[1] https://postimg.org/image/4cwxn91bb/
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
probably NAT traversal kicked in? By default, in SIP profile there's <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> and auto-nat ACL covers "RFC1918 Excluding your local lan". If you use only private addresses in your whole ccommunication, it's better to define an ACL that denies everything and refer to it in "ext-xxx-ip" parameters. On Sat, Jul 2, 2016 at 9:29 AM, Anonim Stefan <fanx07@gmail.com> wrote:
Hello,
I am trying to establish a conference group call using the setup pictured at [1]. Thus, when a user calls to a group extension, all the users in that group are called.
I have set the sip-ip and rtp-ip bind IPs. Signaling works fine, the phones in the group ring. When I answer I even hear the wave playback conference starter/joiner. I can see bindings on the RTP public IP:PORT (using netstat). I can ngrep packets coming in and out for that ports.
The problem I have is that I can hear no media.
I've been debugging and eliminated the possible networking firewall issues using netcat over UDP and some high ports, between public and private RTP ip, in both directions (firewall allows everything).
Thus, I would say that the problem is on the application side. This kind of looks like a scenario where all conference users are muted, even if I don't have any rules like that in the dialplan group extension. The freeswitch ACLs default="allow" and I don't see any "blocked by acl" logs. The logs look fine: "... muxing call legs".
Any ideas what might be the problem?
Thanks in advance, Stefan
[1] https://postimg.org/image/4cwxn91bb/
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thank you Stanislav, I will try. The question I have is why "denies everything" and not "allows everything" for the new ACL?
I might actually be wrong with this, so you better read the docs at https://freeswitch.org/confluence/display/FREESWITCH/NAT On Sun, Jul 3, 2016 at 10:40 AM, Anonim Stefan <fanx07@gmail.com> wrote:
Thank you Stanislav, I will try.
The question I have is why "denies everything" and not "allows everything" for the new ACL?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Thanks for the link. I tried setting "apply-candidate-acl", "apply-inbound-acl" with an ACL allowing the private "172.16.x.x" network, as described here [1] with no success. I tried starting freeswitch with "-nonat" parameter with no success of working RTP. RTP comes with source 172.16.x.x and I am routing it based on a static route on freeswitch machine via public IP gateway which knows to forward it further. [1] https://freeswitch.org/confluence/display/FREESWITCH/ACL Stefan On Mon, Jul 4, 2016 at 12:13 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I might actually be wrong with this, so you better read the docs at https://freeswitch.org/confluence/display/FREESWITCH/NAT
On Sun, Jul 3, 2016 at 10:40 AM, Anonim Stefan <fanx07@gmail.com> wrote:
Thank you Stanislav, I will try.
The question I have is why "denies everything" and not "allows everything" for the new ACL?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Ok, I've figured this out. This is an example of a simple glitch that could take a lot of time to figure out (^_^). The problem was that I was sending a starter_anounce.wav at 48000 sampling rate in the beginning, when the endpoints were negociating a 8000 rate. Apparently this caused the file to be played successfully in the beginning and broke things, folowingly (on the UACs side), making them to not play any more media. This explains why I was seeing packets coming in and out freeswitch machine and no issues on the networking side. Thanks you Stanislav and Mirko for engaging in this discussion. Stefan On Mon, Jul 4, 2016 at 4:56 PM, Anonim Stefan <fanx07@gmail.com> wrote:
Thanks for the link.
I tried setting "apply-candidate-acl", "apply-inbound-acl" with an ACL allowing the private "172.16.x.x" network, as described here [1] with no success. I tried starting freeswitch with "-nonat" parameter with no success of working RTP.
RTP comes with source 172.16.x.x and I am routing it based on a static route on freeswitch machine via public IP gateway which knows to forward it further.
[1] https://freeswitch.org/confluence/display/FREESWITCH/ACL
Stefan
On Mon, Jul 4, 2016 at 12:13 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I might actually be wrong with this, so you better read the docs at https://freeswitch.org/confluence/display/FREESWITCH/NAT
On Sun, Jul 3, 2016 at 10:40 AM, Anonim Stefan <fanx07@gmail.com> wrote:
Thank you Stanislav, I will try.
The question I have is why "denies everything" and not "allows everything" for the new ACL?
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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aaahh the sample rate :) Glad it is fixed, good luck :) On Mon, Jul 4, 2016 at 4:55 PM, Anonim Stefan <fanx07@gmail.com> wrote:
Ok, I've figured this out. This is an example of a simple glitch that could take a lot of time to figure out (^_^).
The problem was that I was sending a starter_anounce.wav at 48000 sampling rate in the beginning, when the endpoints were negociating a 8000 rate. Apparently this caused the file to be played successfully in the beginning and broke things, folowingly (on the UACs side), making them to not play any more media.
This explains why I was seeing packets coming in and out freeswitch machine and no issues on the networking side.
Thanks you Stanislav and Mirko for engaging in this discussion.
Stefan
On Mon, Jul 4, 2016 at 4:56 PM, Anonim Stefan <fanx07@gmail.com> wrote:
Thanks for the link.
I tried setting "apply-candidate-acl", "apply-inbound-acl" with an ACL allowing the private "172.16.x.x" network, as described here [1] with no success. I tried starting freeswitch with "-nonat" parameter with no success of working RTP.
RTP comes with source 172.16.x.x and I am routing it based on a static route on freeswitch machine via public IP gateway which knows to forward it further.
[1] https://freeswitch.org/confluence/display/FREESWITCH/ACL
Stefan
On Mon, Jul 4, 2016 at 12:13 AM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
I might actually be wrong with this, so you better read the docs at https://freeswitch.org/confluence/display/FREESWITCH/NAT
On Sun, Jul 3, 2016 at 10:40 AM, Anonim Stefan <fanx07@gmail.com> wrote:
Thank you Stanislav, I will try.
The question I have is why "denies everything" and not "allows everything" for the new ACL?
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Regards, Mirko
participants (3)
-
Anonim Stefan
-
Mirko Brankovic
-
Stanislav Sinyagin