Hi, all I have a server that has installed the FreeSwitch software, and the server has a public IP addr. My clients are android phones, the CSipSimple softphone runs on them. There are some issues when these phone call each other. When phone A calls phone B, only the phone B can listen the audio coming from phone A, but the phone A cannot. What's more, the phone B always need some delays to listen the audio from phone A. Therefore, I guess I need set a STUN server to deal with the above problem. So I read related web pages, such as http://wiki.freeswitch.org/wiki/NAT_Traversal. But, How to configure the STUN server in FreeSwitch is ambiguous. Then, I tried to make some configurations as below: vars.xml. <X-PRE-PROCESS cmd="set" data="external_rtp_ip=a.b.c.d"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=a.b.c.d"/> internal.xml <param name="ext-rtp-ip" value="$${ external_rtp_ip }"/> <param name="ext-sip-ip" value="$${ external_sip_ip }"/> external.xml <param name="manage-presence" value="true"/> <param name="ext-rtp-ip" value="$${ external_rtp_ip }"/> <param name="ext-sip-ip" value="$${ external_sip_ip }"/> But, I didn't find where could I set the port of STUN server. So I failed to launch the STUN server. Thus, Could you tell me How to solve the one-way audio problem, and How to configure the STUN server in FreeSwitch? Thanks and sorry if my English is poor. Best regards!
If the FS is on a public IP you don't need to configure any of those params they are for dealing with FS itself behind nat. Try running FS the default way and make all the changes on you client. Sometimes disabling all the NAT features on the client actual can make it work better. On Tue, Dec 24, 2013 at 7:59 PM, alex <xhyuer@gmail.com> wrote:
Hi, all
I have a server that has installed the FreeSwitch software, and the server has a public IP addr.
My clients are android phones, the CSipSimple softphone runs on them.
There are some issues when these phone call each other.
When phone A calls phone B, only the phone B can listen the audio coming from phone A, but the phone A cannot. What’s more, the phone B always need some delays to listen the audio from phone A.
Therefore, I guess I need set a STUN server to deal with the above problem. So I read related web pages, such as http://wiki.freeswitch.org/wiki/NAT_Traversal. But, How to configure the STUN server in FreeSwitch is ambiguous.
Then, I tried to make some configurations as below:
vars.xml.
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=a.b.c.d"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=a.b.c.d"/>
internal.xml
<param name="ext-rtp-ip" value="$${ external_rtp_ip }"/>
<param name="ext-sip-ip" value="$${ external_sip_ip }"/>
external.xml
<param name="manage-presence" value="true"/>
<param name="ext-rtp-ip" value="$${ external_rtp_ip }"/>
<param name="ext-sip-ip" value="$${ external_sip_ip }"/>
But, I didn’t find where could I set the port of STUN server. So I failed to launch the STUN server.
Thus, Could you tell me How to solve the one-way audio problem, and How to configure the STUN server in FreeSwitch?
Thanks and sorry if my English is poor.
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
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
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
Have you set STUN in the clients? IIRC, CSipSimple doesn't do Rport, at least it's not an option in the settings, so you also must make sure your firewall opens any rtp ports that FreeSwitch uses. On 27/12/13 03:56, Anthony Minessale wrote:
If the FS is on a public IP you don't need to configure any of those params they are for dealing with FS itself behind nat. Try running FS the default way and make all the changes on you client. Sometimes disabling all the NAT features on the client actual can make it work better.
On Tue, Dec 24, 2013 at 7:59 PM, alex <xhyuer@gmail.com <mailto:xhyuer@gmail.com>> wrote:
Hi, all____
__ __
I have a server that has installed the FreeSwitch software, and the server has a public IP addr.____
__ __
My clients are android phones, the CSipSimple softphone runs on them.____
__ __
There are some issues when these phone call each other.____
__ __
When phone A calls phone B, only the phone B can listen the audio coming from phone A, but the phone A cannot. What’s more, the phone B always need some delays to listen the audio from phone A.____
__ __
Therefore, I guess I need set a STUN server to deal with the above problem. So I read related web pages, such as http://wiki.freeswitch.org/wiki/NAT_Traversal. But, How to configure the STUN server in FreeSwitch is ambiguous.____
__ __
Then, I tried to make some configurations as below:____
__ __
vars.xml.____
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=a.b.c.d"/> ____
<X-PRE-PROCESS cmd="set" data="external_sip_ip=a.b.c.d"/> ____
__ __
internal.xml____
<param name="ext-rtp-ip" value="$${ external_rtp_ip }"/>____
<param name="ext-sip-ip" value="$${ external_sip_ip }"/>____
__ __
external.xml____
<param name="manage-presence" value="true"/>____
<param name="ext-rtp-ip" value="$${ external_rtp_ip }"/>____
<param name="ext-sip-ip" value="$${ external_sip_ip }"/>____
__ __
But, I didn’t find where could I set the port of STUN server. So I failed to launch the STUN server.____
__ __
Thus, Could you tell me How to solve the one-way audio problem, and How to configure the STUN server in FreeSwitch?____
__ __
Thanks and sorry if my English is poor.____
__ __
Best regards!____
_________________________________________________________________________ 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
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
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net <http://irc.freenode.net> #freeswitch ☞ _http://freeswitch.org/g+_
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> ☎ +19193869900
_________________________________________________________________________ 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
I had similar issues with Polycom phones that do not support RPORT. The fix for me was adding these to the directory entry: <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <param name="NDLB-force-rport" value="safe"/> Sean -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Francis Sent: Thursday, December 26, 2013 6:44 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Problems about STUN and one way audio Have you set STUN in the clients? IIRC, CSipSimple doesn't do Rport, at least it's not an option in the settings, so you also must make sure your firewall opens any rtp ports that FreeSwitch uses. On 27/12/13 03:56, Anthony Minessale wrote:
If the FS is on a public IP you don't need to configure any of those params they are for dealing with FS itself behind nat. Try running FS the default way and make all the changes on you client. Sometimes disabling all the NAT features on the client actual can make it work better.
On Tue, Dec 24, 2013 at 7:59 PM, alex <xhyuer@gmail.com <mailto:xhyuer@gmail.com>> wrote:
Hi, all____
__ __
I have a server that has installed the FreeSwitch software, and the server has a public IP addr.____
__ __
My clients are android phones, the CSipSimple softphone runs on them.____
__ __
There are some issues when these phone call each other.____
__ __
When phone A calls phone B, only the phone B can listen the audio coming from phone A, but the phone A cannot. What’s more, the phone B always need some delays to listen the audio from phone A.____
__ __
Therefore, I guess I need set a STUN server to deal with the above problem. So I read related web pages, such as http://wiki.freeswitch.org/wiki/NAT_Traversal. But, How to configure the STUN server in FreeSwitch is ambiguous.____
__ __
Then, I tried to make some configurations as below:____
__ __
vars.xml.____
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=a.b.c.d"/> ____
<X-PRE-PROCESS cmd="set" data="external_sip_ip=a.b.c.d"/> ____
__ __
internal.xml____
<param name="ext-rtp-ip" value="$${ external_rtp_ip }"/>____
<param name="ext-sip-ip" value="$${ external_sip_ip }"/>____
__ __
external.xml____
<param name="manage-presence" value="true"/>____
<param name="ext-rtp-ip" value="$${ external_rtp_ip }"/>____
<param name="ext-sip-ip" value="$${ external_sip_ip }"/>____
__ __
But, I didn’t find where could I set the port of STUN server. So I failed to launch the STUN server.____
__ __
Thus, Could you tell me How to solve the one-way audio problem, and How to configure the STUN server in FreeSwitch?____
__ __
Thanks and sorry if my English is poor.____
__ __
Best regards!____
_________________________________________________________________________ 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
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
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net <http://irc.freenode.net> #freeswitch ☞ _http://freeswitch.org/g+_
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> ☎ +19193869900
______________________________________________________________________ ___ 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-use rs 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
safe only applies rport to Polycom user agents… along with the KIRK series. -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Dec 27, 2013, at 12:34 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
I had similar issues with Polycom phones that do not support RPORT. The fix for me was adding these to the directory entry:
<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <param name="NDLB-force-rport" value="safe"/>
Sean
participants (5)
-
alex
-
Anthony Minessale
-
Brian West
-
Francis
-
Sean Devoy