Oneway audio issues in freeswitch
Hello All We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server. Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine) We have tried multiple softphones and the result is same. Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch. Any help or hint will be much appreciated. Thank you,
You need to look at the sip signaling to see what's going on On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v@gmail.com> wrote:
Hello All
We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server.
Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine)
We have tried multiple softphones and the result is same.
Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch.
Any help or hint will be much appreciated.
Thank you,
_________________________________________________________________________ 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
-- Ítalo Rossi italo@freeswitch.org
As a rule of dumb, try turning on rport Le 10 févr. 2016 8:55 PM, "Ítalo Rossi" <italo@freeswitch.org> a écrit :
You need to look at the sip signaling to see what's going on
On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v@gmail.com> wrote:
Hello All
We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server.
Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine)
We have tried multiple softphones and the result is same.
Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch.
Any help or hint will be much appreciated.
Thank you,
_________________________________________________________________________ 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
-- Ítalo Rossi italo@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
Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz < luis.daniel.lucio@gmail.com> wrote:
As a rule of dumb, try turning on rport Le 10 févr. 2016 8:55 PM, "Ítalo Rossi" <italo@freeswitch.org> a écrit :
You need to look at the sip signaling to see what's going on
On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v@gmail.com> wrote:
Hello All
We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server.
Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine)
We have tried multiple softphones and the result is same.
Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch.
Any help or hint will be much appreciated.
Thank you,
_________________________________________________________________________ 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
-- Ítalo Rossi italo@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
_________________________________________________________________________ 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
without logs of a call doing this at debug level with a complete unmolested sip trace in line its a little hard to speculate whats going on here On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque <shafeeq.v@gmail.com> wrote:
Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x
On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz < luis.daniel.lucio@gmail.com> wrote:
As a rule of dumb, try turning on rport Le 10 févr. 2016 8:55 PM, "Ítalo Rossi" <italo@freeswitch.org> a écrit :
You need to look at the sip signaling to see what's going on
On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v@gmail.com
wrote:
Hello All
We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server.
Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine)
We have tried multiple softphones and the result is same.
Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch.
Any help or hint will be much appreciated.
Thank you,
_________________________________________________________________________ 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
-- Ítalo Rossi italo@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
_________________________________________________________________________ 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
Hello Ken, Thank for look in this. Attached are debug logs. SIP Traces were not molested, except the public IPs were changed. As of writing of this email, the issue is isolated to 1.6.x. Not sure if anyone else has tested this on latest version. But easy to reproduce. Just download grandstream Wave, available to IOS and Andriod and try to call any extension directly. Curious to see if any one can come with different result. *Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F 949-269-0449 / 949-232-1410 | jprangi@didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | On Thu, Feb 11, 2016 at 8:36 AM, Ken Rice <krice@freeswitch.org> wrote:
without logs of a call doing this at debug level with a complete unmolested sip trace in line its a little hard to speculate whats going on here
On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque <shafeeq.v@gmail.com> wrote:
Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x
On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz < luis.daniel.lucio@gmail.com> wrote:
As a rule of dumb, try turning on rport Le 10 févr. 2016 8:55 PM, "Ítalo Rossi" <italo@freeswitch.org> a écrit :
You need to look at the sip signaling to see what's going on
On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque < shafeeq.v@gmail.com> wrote:
Hello All
We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server.
Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine)
We have tried multiple softphones and the result is same.
Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch.
Any help or hint will be much appreciated.
Thank you,
_________________________________________________________________________ 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
-- Ítalo Rossi italo@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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
Found the problem, We had this enabled in profiles. auto-jitterbuffer-msec value=60 Commenting it seems to have fixed the issue. Not sure why this would cause problem only from one type of phones. Any hint. Thank you, *Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F 949-269-0449 / 949-232-1410 | jprangi@didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | On Sat, Feb 13, 2016 at 9:45 AM, Jai Rangi <jprangi@didforsale.com> wrote:
Hello Ken, Thank for look in this. Attached are debug logs. SIP Traces were not molested, except the public IPs were changed. As of writing of this email, the issue is isolated to 1.6.x. Not sure if anyone else has tested this on latest version. But easy to reproduce. Just download grandstream Wave, available to IOS and Andriod and try to call any extension directly. Curious to see if any one can come with different result.
*Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F 949-269-0449 / 949-232-1410 | jprangi@didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 |
On Thu, Feb 11, 2016 at 8:36 AM, Ken Rice <krice@freeswitch.org> wrote:
without logs of a call doing this at debug level with a complete unmolested sip trace in line its a little hard to speculate whats going on here
On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque <shafeeq.v@gmail.com
wrote:
Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x
On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz < luis.daniel.lucio@gmail.com> wrote:
As a rule of dumb, try turning on rport Le 10 févr. 2016 8:55 PM, "Ítalo Rossi" <italo@freeswitch.org> a écrit :
You need to look at the sip signaling to see what's going on
On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque < shafeeq.v@gmail.com> wrote:
Hello All
We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server.
Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine)
We have tried multiple softphones and the result is same.
Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch.
Any help or hint will be much appreciated.
Thank you,
_________________________________________________________________________ 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
-- Ítalo Rossi italo@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
_________________________________________________________________________ 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
_________________________________________________________________________ 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
Redo with sofia global siptrace on so we can see the SIP messaging On Feb 13, 2016, at 12:45 PM, Jai Rangi <jprangi@didforsale.com> wrote: Hello Ken, Thank for look in this. Attached are debug logs. SIP Traces were not molested, except the public IPs were changed. As of writing of this email, the issue is isolated to 1.6.x. Not sure if anyone else has tested this on latest version. But easy to reproduce. Just download grandstream Wave, available to IOS and Andriod and try to call any extension directly. Curious to see if any one can come with different result. Jai Rangi Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 <tel:O%20949-471-0101> | C 949-419-7634 <tel:C%20949-742-2666> | F 949-269-0449 / 949-232-1410 <> | jprangi@didforsale.com <mailto:jprangi@didforsale.com> www.cebod.com <http://www.cebod.com/> | www.didforsale.com <http://www.didforsale.com/> |3200 Bristol St Suite 615, Costa Mesa, CA 92626 | On Thu, Feb 11, 2016 at 8:36 AM, Ken Rice <krice@freeswitch.org <mailto:krice@freeswitch.org>> wrote: without logs of a call doing this at debug level with a complete unmolested sip trace in line its a little hard to speculate whats going on here On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque <shafeeq.v@gmail.com <mailto:shafeeq.v@gmail.com>> wrote: Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz <luis.daniel.lucio@gmail.com <mailto:luis.daniel.lucio@gmail.com>> wrote: As a rule of dumb, try turning on rport Le 10 févr. 2016 8:55 PM, "Ítalo Rossi" <italo@freeswitch.org <mailto:italo@freeswitch.org>> a écrit : You need to look at the sip signaling to see what's going on On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v@gmail.com <mailto:shafeeq.v@gmail.com>> wrote: Hello All We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server. Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine) We have tried multiple softphones and the result is same. Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch. Any help or hint will be much appreciated. Thank you, _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> -- Ítalo Rossi italo@freeswitch.org <mailto:italo@freeswitch.org> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <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 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> <freeswitch.log.goodcall.txt><freeswitch.log.badcall.txt>_________________________________________________________________________ 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 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Hello Dave, Sorry for the delay, have been out of office. Attached the the debug log file, with siptrace on. This is coming from default configuration. I can confirm the issue happens only when this line is commented out. <param name="auto-jitterbuffer-msec" value="60"/> Not sure if this qualifies for bug, I can open a ticket in Jira. Thank you, -Jai On Sat, Feb 13, 2016 at 7:49 PM, Dave Horton <dave@dchorton.com> wrote:
Redo with sofia global siptrace on so we can see the SIP messaging On Feb 13, 2016, at 12:45 PM, Jai Rangi <jprangi@didforsale.com> wrote:
Hello Ken, Thank for look in this. Attached are debug logs. SIP Traces were not molested, except the public IPs were changed. As of writing of this email, the issue is isolated to 1.6.x. Not sure if anyone else has tested this on latest version. But easy to reproduce. Just download grandstream Wave, available to IOS and Andriod and try to call any extension directly. Curious to see if any one can come with different result.
*Jai Rangi* Cebod Technologies LLC dba DIDforSale/Cebod Telecom O 949-471-0102 <O%20949-471-0101> | C 949-419-7634 <C%20949-742-2666> | F 949-269-0449 / 949-232-1410 | jprangi@didforsale.com www.cebod.com | www.didforsale.com |3200 Bristol St Suite 615, Costa Mesa, CA 92626 |
On Thu, Feb 11, 2016 at 8:36 AM, Ken Rice <krice@freeswitch.org> wrote:
without logs of a call doing this at debug level with a complete unmolested sip trace in line its a little hard to speculate whats going on here
On Thu, Feb 11, 2016 at 10:28 AM, mohammed shafeeque <shafeeq.v@gmail.com
wrote:
Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x
On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz < luis.daniel.lucio@gmail.com> wrote:
As a rule of dumb, try turning on rport Le 10 févr. 2016 8:55 PM, "Ítalo Rossi" <italo@freeswitch.org> a écrit :
You need to look at the sip signaling to see what's going on
On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque < shafeeq.v@gmail.com> wrote:
Hello All
We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server.
Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine)
We have tried multiple softphones and the result is same.
Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch.
Any help or hint will be much appreciated.
Thank you,
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-- Ítalo Rossi italo@freeswitch.org
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<freeswitch.log.goodcall.txt><freeswitch.log.badcall.txt> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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We have been looking at that all day, but cant figure out the issue. Funny thing is that its happening only when GS Originate the call. May be we are over looking something. Here are two call example. IPs are modified for security. FreeSWITCH (Version 1.6.6 64bit) is ready freeswitch@internal> sofia_contact 1276@domain.company.com sofia/internal/sip:1276@192.168.1.168:12113 ;fs_nat=yes;fs_path=sip%3A1276%4068.5.194.163%3A12113 freeswitch@internal> sofia_contact 142@domain.company.com sofia/internal/sip:142@172.16.42.13:11852 ;fs_nat=yes;fs_path=sip%3A142%4074.67.200.39%3A33812 142 calls 1276 (1276 does not hear anything) (Seems freeswitch not handling nat properly) On TCP dump, I can see free switching receiving the RTP packet, but trying to deliver to local IP for 1276. 1276 calls 142 (All good both parties can hear) 142 call PSTN number (All good) 1272 call PSTN number (All good) Same configuration, same dialplan works just fine with 1.6.2 and 1.4.18. 1.6.2 had intermittent DTMF issue, we upgraded to 1.6.5, found this one way audio, upgraded to 1.6.6. We have narrowed it down to Grand Stream Softphone and Grad Stream IP phones. Here is trace for Both calls. ###################################### U 74.67.200.39:33812 -> 222.222.222.222:5060 INVITE sip:1276@domain.company.com:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 140 INVITE. Contact: "142" <sip:142@172.16.42.13:11852>. Max-Forwards: 70. User-Agent: Grandstream Wave/IOS 1.0.19. Privacy: none. P-Preferred-Identity: "142" <sip:142@domain.company.com:5060>. Supported: replaces, path, timer, eventlist. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 235. . v=0. o=142 8000 8000 IN IP4 172.16.42.13. s=SIP Call. c=IN IP4 172.16.42.13. t=0 0. m=audio 50476 RTP/AVP 0 8 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK1892445982;rport=33812;received=74.67.200.39. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 140 INVITE. User-Agent: VOIPGATEWAY. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK1892445982;rport=33812;received=74.67.200.39. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>;tag=Kt2jU0QN57N8F. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 140 INVITE. User-Agent: VOIPGATEWAY. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Proxy-Authenticate: Digest realm="domain.company.com", nonce="5eb04922-17e3-426e-9ddf-4333692102b5", algorithm=MD5, qop="auth". Content-Length: 0. . ## U 74.67.200.39:33812 -> 222.222.222.222:5060 ACK sip:1276@domain.company.com:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1892445982;rport. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>;tag=Kt2jU0QN57N8F. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 140 ACK. Content-Length: 0. . # U 74.67.200.39:33812 -> 222.222.222.222:5060 INVITE sip:1276@domain.company.com:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK322043518;rport. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 141 INVITE. Contact: "142" <sip:142@172.16.42.13:11852>. Proxy-Authorization: Digest username="xxxxx", realm="domain.company.com", nonce="5eb04922-17e3-426e-9ddf-4333692102b5", uri=" sip:1276@domain.company.com:5060", response="d309cb76b83042023f6794835ad60a89", algorithm=MD5, cnonce="15121380", qop=auth, nc=00000003. Max-Forwards: 70. User-Agent: Grandstream Wave/IOS 1.0.19. Privacy: none. P-Preferred-Identity: "142" <sip:142@domain.company.com:5060>. Supported: replaces, path, timer, eventlist. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 235. . v=0. o=142 8000 8000 IN IP4 172.16.42.13. s=SIP Call. c=IN IP4 172.16.42.13. t=0 0. m=audio 50476 RTP/AVP 0 8 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 141 INVITE. User-Agent: VOIPGATEWAY. Content-Length: 0. . # U 222.222.222.222:5060 -> 68.5.194.163:12113 INVITE sip:1276@192.168.1.168:12113 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKX5H9FDZX1eyNN. Route: <sip:1276@68.5.194.163:12113>. Max-Forwards: 68. From: "Softphone" <sip:142@domain.company.com>;tag=NcN4ypSvZS2DQ. To: <sip:1276@192.168.1.168:12113>. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243919 INVITE. Contact: <sip:142@222.222.222.222:5060>. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 270. X-FS-Support: update_display,send_info. . v=0. o=FreeSWITCH 1455142324 1455142325 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 17642 RTP/AVP 0 8 101 13. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. # # U 68.5.194.163:12113 -> 222.222.222.222:5060 SIP/2.0 100 Trying. From: Softphone<sip:142@domain.company.com>;tag=NcN4ypSvZS2DQ. To: sip:1276@192.168.1.168:12113. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243919 INVITE. Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN. Content-Length: 0. . ### U 68.5.194.163:12113 -> 222.222.222.222:5060 SIP/2.0 180 Ringing. From: Softphone<sip:142@domain.company.com>;tag=NcN4ypSvZS2DQ. To: sip:1276@192.168.1.168:12113;tag=ReKf2-xIWCt. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243919 INVITE. Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN. Contact: ATAPHONE<sip:1276@192.168.1.168:12113>. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>;tag=m3UBXU8r2gcUB. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 141 INVITE. Contact: <sip:1276@222.222.222.222:5060;transport=udp>. User-Agent: VOIPGATEWAY. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 222. P-Asserted-Identity: "1276" <sip:1276@domain.company.com>. . v=0. o=FreeSWITCH 1455132056 1455132057 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 27910 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. ############## ## ####### U 68.5.194.163:12113 -> 222.222.222.222:5060 SIP/2.0 200 OK. From: Softphone<sip:142@domain.company.com>;tag=NcN4ypSvZS2DQ. To: sip:1276@192.168.1.168:12113;tag=ReKf2-xIWCt. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243919 INVITE. Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bKX5H9FDZX1eyNN. Contact: ATAPHONE<sip:1276@192.168.1.168:12113>. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER. Supported: timer,replaces. Content-Type: application/sdp. Content-Length: 207. . v=0. o=1276 87748 1 IN IP4 192.168.1.168. s=-. c=IN IP4 192.168.1.168. t=0 0. m=audio 8000 RTP/AVP 0 96. a=rtpmap:0 PCMU/8000. a=rtpmap:96 telephone-event/8000. a=ptime:20. a=rtpmap:96 telephone-event/8000. # U 222.222.222.222:5060 -> 68.5.194.163:12113 ACK sip:1276@192.168.1.168:12113 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKZQ4tK304U0aUc. Max-Forwards: 70. From: "Softphone" <sip:142@domain.company.com>;tag=NcN4ypSvZS2DQ. To: <sip:1276@192.168.1.168:12113>;tag=ReKf2-xIWCt. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243919 ACK. Contact: <sip:142@222.222.222.222:5060>. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK322043518;rport=33812;received=74.67.200.39. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>;tag=m3UBXU8r2gcUB. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 141 INVITE. Contact: <sip:1276@222.222.222.222:5060;transport=udp>. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 222. P-Asserted-Identity: "Outbound Call" <sip:1276@domain.company.com>. . v=0. o=FreeSWITCH 1455132056 1455132057 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 27910 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. ## U 74.67.200.39:33812 -> 222.222.222.222:5060 ACK sip:1276@222.222.222.222:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK1657841580;rport. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>;tag=m3UBXU8r2gcUB. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 141 ACK. Contact: <sip:142@172.16.42.13:11852>. Proxy-Authorization: Digest username="xxxx", realm="domain.company.com", nonce="5eb04922-17e3-426e-9ddf-4333692102b5", uri=" sip:1276@domain.company.com:5060", response="d309cb76b83042023f6794835ad60a89", algorithm=MD5, cnonce="15121380", qop=auth, nc=00000003. Max-Forwards: 70. Supported: replaces, path, timer, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . ############################################################### #### # # ######### ################################################################################################ U 74.67.200.39:33812 -> 222.222.222.222:5060 BYE sip:1276@222.222.222.222:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK326133779;rport. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>;tag=m3UBXU8r2gcUB. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 142 BYE. Contact: <sip:142@172.16.42.13:11852>. Max-Forwards: 70. Supported: replaces, path, timer, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK326133779;rport=33812;received=74.67.200.39. From: "142" <sip:142@domain.company.com:5060>;tag=479799221. To: <sip:1276@domain.company.com:5060>;tag=m3UBXU8r2gcUB. Call-ID: 152039027-11852-15@BHC.BG.EC.BD. CSeq: 142 BYE. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Content-Length: 0. . ### U 222.222.222.222:5060 -> 68.5.194.163:12113 BYE sip:1276@192.168.1.168:12113 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK2jg5rmKFKUDKF. Max-Forwards: 70. From: "Softphone" <sip:142@domain.company.com>;tag=NcN4ypSvZS2DQ. To: <sip:1276@192.168.1.168:12113>;tag=ReKf2-xIWCt. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243920 BYE. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. . ### U 68.5.194.163:12113 -> 222.222.222.222:5060 SIP/2.0 200 OK. From: Softphone<sip:142@domain.company.com>;tag=NcN4ypSvZS2DQ. To: sip:1276@192.168.1.168:12113;tag=ReKf2-xIWCt. Call-ID: 3ba9d1ca-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243920 BYE. Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK2jg5rmKFKUDKF. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Content-Length: 0. . ####################################################### ######### # Successful Call with 2 way audio :1276 calls 142 # # # ################################## U 68.5.194.163:12113 -> 222.222.222.222:5060 INVITE sip:142@domain.company.com SIP/2.0. From: ATAPHONE<sip:1276@domain.company.com>;tag=VxLf2-dmKe50. To: sip:142@domain.company.com. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 176 INVITE. Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH. Contact: ATAPHONE<sip:1276@192.168.1.168:12113>. Max-Forwards: 70. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER. Supported: timer,replaces. Content-Type: application/sdp. Content-Length: 257. . v=0. o=1276 87384 1 IN IP4 192.168.1.168. s=-. c=IN IP4 192.168.1.168. t=0 0. m=audio 8002 RTP/AVP 0 18 8 96. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 telephone-event/8000. a=ptime:20. a=rtpmap:96 telephone-event/8000. # U 222.222.222.222:5060 -> 68.5.194.163:12113 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.168:12113 ;branch=z9hG4bKu22f2-74MJFSKH;received=68.5.194.163. From: ATAPHONE <sip:1276@domain.company.com>;tag=VxLf2-dmKe50. To: <sip:142@domain.company.com>. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 176 INVITE. User-Agent: VOIPGATEWAY. Content-Length: 0. . # U 222.222.222.222:5060 -> 68.5.194.163:12113 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.1.168:12113 ;branch=z9hG4bKu22f2-74MJFSKH;received=68.5.194.163. From: ATAPHONE <sip:1276@domain.company.com>;tag=VxLf2-dmKe50. To: <sip:142@domain.company.com>;tag=SjctQ6jcUQD5a. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 176 INVITE. User-Agent: VOIPGATEWAY. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Proxy-Authenticate: Digest realm="domain.company.com", nonce="1cf240ca-c5d1-4ac0-8417-becd41784ffb", algorithm=MD5, qop="auth". Content-Length: 0. . # U 68.5.194.163:12113 -> 222.222.222.222:5060 ACK sip:142@domain.company.com SIP/2.0. From: ATAPHONE<sip:1276@domain.company.com>;tag=VxLf2-dmKe50. To: sip:142@domain.company.com;tag=SjctQ6jcUQD5a. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 176 ACK. Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-74MJFSKH. Content-Length: 0. . # U 68.5.194.163:12113 -> 222.222.222.222:5060 INVITE sip:142@domain.company.com SIP/2.0. From: ATAPHONE<sip:1276@domain.company.com>;tag=VxLf2-dmKe50. To: sip:142@domain.company.com. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 177 INVITE. Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-l5MoOcq0. Contact: ATAPHONE<sip:1276@192.168.1.168:12113>. Max-Forwards: 70. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER. Supported: timer,replaces. Proxy-Authorization: Digest username="yyyy",realm="domain.company.com",uri=" sip:142@domain.company.com ",response="a87ca168b1869994a6b4782df7bffe99",algorithm=MD5,nonce="1cf240ca-c5d1-4ac0-8417-becd41784ffb",qop=auth,cnonce="00017214",nc=00000001. Content-Type: application/sdp. Content-Length: 257. . v=0. o=1276 87384 1 IN IP4 192.168.1.168. s=-. c=IN IP4 192.168.1.168. t=0 0. m=audio 8002 RTP/AVP 0 18 8 96. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 telephone-event/8000. a=ptime:20. a=rtpmap:96 telephone-event/8000. # U 222.222.222.222:5060 -> 68.5.194.163:12113 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.168:12113 ;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163. From: ATAPHONE <sip:1276@domain.company.com>;tag=VxLf2-dmKe50. To: <sip:142@domain.company.com>. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 177 INVITE. User-Agent: VOIPGATEWAY. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 INVITE sip:142@172.16.42.13:11852 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK6pN8Z0pX7y6XD. Route: <sip:142@74.67.200.39:33812>. Max-Forwards: 68. From: "ATA" <sip:1276@domain.company.com>;tag=U4yBUvmKN9Saj. To: <sip:142@172.16.42.13:11852>. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 INVITE. Contact: <sip:1276@222.222.222.222:5060>. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 270. X-FS-Support: update_display,send_info. . v=0. o=FreeSWITCH 1455141647 1455141648 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 18346 RTP/AVP 0 8 101 13. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. # U 222.222.222.222:5060 -> 74.67.200.39:33812 INVITE sip:142@172.16.42.13:11852 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK6pN8Z0pX7y6XD. Route: <sip:142@74.67.200.39:33812>. Max-Forwards: 68. From: "ATA" <sip:1276@domain.company.com>;tag=U4yBUvmKN9Saj. To: <sip:142@172.16.42.13:11852>. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 INVITE. Contact: <sip:1276@222.222.222.222:5060>. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 270. X-FS-Support: update_display,send_info. . v=0. o=FreeSWITCH 1455141647 1455141648 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 18346 RTP/AVP 0 8 101 13. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20. ## U 74.67.200.39:33812 -> 222.222.222.222:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 222.222.222.222:5060 ;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD. From: "ATA" <sip:1276@domain.company.com>;tag=U4yBUvmKN9Saj. To: <sip:142@172.16.42.13:11852>. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 INVITE. Supported: replaces, path, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . # U 74.67.200.39:33812 -> 222.222.222.222:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 222.222.222.222:5060 ;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD. From: "ATA" <sip:1276@domain.company.com>;tag=U4yBUvmKN9Saj. To: <sip:142@172.16.42.13:11852>;tag=1713531477. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 INVITE. Contact: <sip:142@172.16.42.13:11852>. Supported: replaces, path, timer, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow-Events: talk, hold. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . # U 222.222.222.222:5060 -> 68.5.194.163:12113 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 192.168.1.168:12113 ;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163. From: ATAPHONE <sip:1276@domain.company.com>;tag=VxLf2-dmKe50. To: <sip:142@domain.company.com>;tag=tU5jS13Fr03Qp. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 177 INVITE. Contact: <sip:142@222.222.222.222:5060;transport=udp>. User-Agent: VOIPGATEWAY. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 219. Remote-Party-ID: "142" <sip:142@domain.company.com
;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1455140898 1455140899 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 19096 RTP/AVP 0 96. a=rtpmap:0 PCMU/8000. a=rtpmap:96 telephone-event/8000. a=fmtp:96 0-16. a=ptime:20.
################################################################## U 74.67.200.39:33812 -> 222.222.222.222:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 222.222.222.222:5060 ;rport=5660;branch=z9hG4bK6pN8Z0pX7y6XD. From: "ATA" <sip:1276@domain.company.com>;tag=U4yBUvmKN9Saj. To: <sip:142@172.16.42.13:11852>;tag=1713531477. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 INVITE. Contact: <sip:142@172.16.42.13:11852>. Supported: replaces, path, timer, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Content-Length: 235. . v=0. o=142 8000 8000 IN IP4 172.16.42.13. s=SIP Call. c=IN IP4 172.16.42.13. t=0 0. m=audio 26390 RTP/AVP 0 8 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 222.222.222.222:5060 -> 74.67.200.39:33812 ACK sip:142@172.16.42.13:11852 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bKB4K487aFSBpUQ. Max-Forwards: 70. From: "ATA" <sip:1276@domain.company.com>;tag=U4yBUvmKN9Saj. To: <sip:142@172.16.42.13:11852>;tag=1713531477. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243932 ACK. Contact: <sip:1276@222.222.222.222:5060>. Content-Length: 0. . # U 222.222.222.222:5060 -> 68.5.194.163:12113 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.168:12113 ;branch=z9hG4bKu22f2-l5MoOcq0;received=68.5.194.163. From: ATAPHONE <sip:1276@domain.company.com>;tag=VxLf2-dmKe50. To: <sip:142@domain.company.com>;tag=tU5jS13Fr03Qp. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 177 INVITE. Contact: <sip:142@222.222.222.222:5060;transport=udp>. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 219. Remote-Party-ID: "Outbound Call" <sip:142@domain.company.com
;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1455140898 1455140899 IN IP4 222.222.222.222. s=FreeSWITCH. c=IN IP4 222.222.222.222. t=0 0. m=audio 19096 RTP/AVP 0 96. a=rtpmap:0 PCMU/8000. a=rtpmap:96 telephone-event/8000. a=fmtp:96 0-16. a=ptime:20.
# U 68.5.194.163:12113 -> 222.222.222.222:5060 ACK sip:142@222.222.222.222:5060 SIP/2.0. From: ATAPHONE<sip:1276@domain.company.com>;tag=VxLf2-dmKe50. To: sip:142@domain.company.com;tag=tU5jS13Fr03Qp. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 177 ACK. Via: SIP/2.0/UDP 192.168.1.168:12113;branch=z9hG4bKu22f2-VwM5aez*. Contact: ATAPHONE<sip:1276@192.168.1.168:12113>. Max-Forwards: 70. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Content-Length: 0. . ########################################################################################### ############################## ############################################################################################################################################################## U 74.67.200.39:33812 -> 222.222.222.222:5060 BYE sip:1276@222.222.222.222:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.42.13:11852;branch=z9hG4bK364522226;rport. From: <sip:142@172.16.42.13:11852>;tag=1713531477. To: "ATA" <sip:1276@domain.company.com>;tag=U4yBUvmKN9Saj. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243933 BYE. Contact: <sip:142@172.16.42.13:11852>. Max-Forwards: 70. Supported: replaces, path, timer, eventlist. User-Agent: Grandstream Wave/IOS 1.0.19. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . # U 222.222.222.222:5060 -> 74.67.200.39:33812 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.42.13:11852 ;branch=z9hG4bK364522226;rport=33812;received=74.67.200.39. From: <sip:142@172.16.42.13:11852>;tag=1713531477. To: "ATA" <sip:1276@domain.company.com>;tag=U4yBUvmKN9Saj. Call-ID: 4bf201f9-4b0f-1234-88a0-00a0d1eb190c. CSeq: 87243933 BYE. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Content-Length: 0. . # U 222.222.222.222:5060 -> 68.5.194.163:12113 BYE sip:1276@192.168.1.168:12113 SIP/2.0. Via: SIP/2.0/UDP 222.222.222.222:5060;rport;branch=z9hG4bK89gKvrKN5DvSQ. Max-Forwards: 70. From: <sip:142@domain.company.com>;tag=tU5jS13Fr03Qp. To: ATAPHONE <sip:1276@domain.company.com>;tag=VxLf2-dmKe50. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 87243943 BYE. User-Agent: VOIPGATEWAY. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. ## U 68.5.194.163:12113 -> 222.222.222.222:5060 SIP/2.0 200 OK. From: sip:142@domain.company.com;tag=tU5jS13Fr03Qp. To: ATAPHONE<sip:1276@domain.company.com>;tag=VxLf2-dmKe50. Call-ID: d44p80-lYfd5f2@domain.company.com. CSeq: 87243943 BYE. Via: SIP/2.0/UDP 222.222.222.222:5060;branch=z9hG4bK89gKvrKN5DvSQ. User-Agent: Patton Smartlink MATA <4.02.0A1 MS VR CS (0001)><00a0ba0a12e5>. Content-Length: 0. On Wed, Feb 10, 2016 at 5:54 PM, Ítalo Rossi <italo@freeswitch.org> wrote:
You need to look at the sip signaling to see what's going on
On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v@gmail.com> wrote:
Hello All
We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server.
Here is scenario: Grandstream call any extensions (one way audio) Any extension call Grandstream ( Audio works just fine)
We have tried multiple softphones and the result is same.
Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch.
Any help or hint will be much appreciated.
Thank you,
_________________________________________________________________________ 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
-- Ítalo Rossi italo@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
participants (7)
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Dave Horton
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Jai Rangi
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Jai Rangi
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Ken Rice
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Luis Daniel Lucio Quiroz
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mohammed shafeeque
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Ítalo Rossi