No audio on either of an established call
I currently have 2 SIP clients (Linphone) successfully calling each other, but there is no audio on either end. The set up is as follows: Linphone1 (1000) --> Kamailio 1 <-------> Freeswitch <------> Kamalio 2 <---- Linphone2 (2000) Using Freeswitch 1.2.12 on CentOS (installed via RPM) Freeswitch has two interfaces: external - 10.1.1.206 internal - 10.10.10.206 Each of the Linphone clients are registered their respective Kamailio instance and Kamailio is configured to route via the appropriate interface on Freeswitch. The SIP negotiation is working as I can call either Linphone client. I've done a tcpdump on each side of Freeswitch and can see the RTP traffic between the Linphone and the appropriate interface on Freeswitch. I've tried different codec combinations (mostly G711 and iLBC), and different SIP clients but still get no audio. Any pointers on how to track down the issue? Thanks Peter
I'd start by getting a debug log of the call and posting it on pastebin.freeswitch.org. Hopefully that will yield some clues. -MC On Wed, Jul 31, 2013 at 7:33 PM, Peter <eidevm5@gmail.com> wrote:
I currently have 2 SIP clients (Linphone) successfully calling each other, but there is no audio on either end.
The set up is as follows:
Linphone1 (1000) --> Kamailio 1 <-------> Freeswitch <------> Kamalio 2 <---- Linphone2 (2000)
Using Freeswitch 1.2.12 on CentOS (installed via RPM)
Freeswitch has two interfaces:
external - 10.1.1.206 internal - 10.10.10.206
Each of the Linphone clients are registered their respective Kamailio instance and Kamailio is configured to route via the appropriate interface on Freeswitch.
The SIP negotiation is working as I can call either Linphone client.
I've done a tcpdump on each side of Freeswitch and can see the RTP traffic between the Linphone and the appropriate interface on Freeswitch.
I've tried different codec combinations (mostly G711 and iLBC), and different SIP clients but still get no audio.
Any pointers on how to track down the issue?
Thanks
Peter
_________________________________________________________________________ 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
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Here's the Freeswitch log showing the SIP traffic: http://pastebin.freeswitch.org/21253 The summary of IP addresses are: 10.1.1.19 - External Kamailio 10.1.1.206 - External FS interface 10.10.10.206 - Internal FS interface 10.10.10.207 - Internal Kamailio Call was made from 2005 (10.1.254.25) to 1008 (10.10.10.165). Note that 10.1.254.0/24 is routable to 10.1.1.0/24 Here's the tcpdump sample taken on both interfaces on Freeswitch http://pastebin.freeswitch.org/21254 Let me know if anymore logs/debugs are required. On Thu, Aug 1, 2013 at 1:42 PM, Michael Collins <msc@freeswitch.org> wrote:
I'd start by getting a debug log of the call and posting it on pastebin.freeswitch.org. Hopefully that will yield some clues. -MC
On Wed, Jul 31, 2013 at 7:33 PM, Peter <eidevm5@gmail.com> wrote:
I currently have 2 SIP clients (Linphone) successfully calling each other, but there is no audio on either end.
The set up is as follows:
Linphone1 (1000) --> Kamailio 1 <-------> Freeswitch <------> Kamalio 2 <---- Linphone2 (2000)
Using Freeswitch 1.2.12 on CentOS (installed via RPM)
Freeswitch has two interfaces:
external - 10.1.1.206 internal - 10.10.10.206
Each of the Linphone clients are registered their respective Kamailio instance and Kamailio is configured to route via the appropriate interface on Freeswitch.
The SIP negotiation is working as I can call either Linphone client.
I've done a tcpdump on each side of Freeswitch and can see the RTP traffic between the Linphone and the appropriate interface on Freeswitch.
I've tried different codec combinations (mostly G711 and iLBC), and different SIP clients but still get no audio.
Any pointers on how to track down the issue?
Thanks
Peter
_________________________________________________________________________ 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
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.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
After checking everything for the umpteenth time, I discovered I was missing a route back to 10.1.254.0/24 Once I put that in, bingo, audio! On Thu, Aug 1, 2013 at 12:33 PM, Peter <eidevm5@gmail.com> wrote:
I currently have 2 SIP clients (Linphone) successfully calling each other, but there is no audio on either end.
The set up is as follows:
Linphone1 (1000) --> Kamailio 1 <-------> Freeswitch <------> Kamalio 2 <---- Linphone2 (2000)
Using Freeswitch 1.2.12 on CentOS (installed via RPM)
Freeswitch has two interfaces:
external - 10.1.1.206 internal - 10.10.10.206
Each of the Linphone clients are registered their respective Kamailio instance and Kamailio is configured to route via the appropriate interface on Freeswitch.
The SIP negotiation is working as I can call either Linphone client.
I've done a tcpdump on each side of Freeswitch and can see the RTP traffic between the Linphone and the appropriate interface on Freeswitch.
I've tried different codec combinations (mostly G711 and iLBC), and different SIP clients but still get no audio.
Any pointers on how to track down the issue?
Thanks
Peter
Where on your diagram is the Router? I assume that your Kamailio 1 and Kamailio 2 are on two different networks? Paul On Wed, 31 Jul, 2013 at 10:12 PM, Peter <eidevm5@gmail.com> wrote:
After checking everything for the umpteenth time, I discovered I was missing a route back to 10.1.254.0/24
Once I put that in, bingo, audio!
On Thu, Aug 1, 2013 at 12:33 PM, Peter <eidevm5@gmail.com> wrote:
I currently have 2 SIP clients (Linphone) successfully calling each other, but there is no audio on either end.
The set up is as follows:
Linphone1 (1000) --> Kamailio 1 <-------> Freeswitch <------> Kamalio 2 <---- Linphone2 (2000)
Using Freeswitch 1.2.12 on CentOS (installed via RPM)
Freeswitch has two interfaces:
external - 10.1.1.206 internal - 10.10.10.206
Each of the Linphone clients are registered their respective Kamailio instance and Kamailio is configured to route via the appropriate interface on Freeswitch.
The SIP negotiation is working as I can call either Linphone client.
I've done a tcpdump on each side of Freeswitch and can see the RTP traffic between the Linphone and the appropriate interface on Freeswitch.
I've tried different codec combinations (mostly G711 and iLBC), and different SIP clients but still get no audio.
Any pointers on how to track down the issue?
Thanks
Peter
Correct. Both Kamailio servers are on different networks. I didn't show the router in the mix for simplicity. On Fri, Aug 2, 2013 at 6:31 AM, Paul <pasha@prosperity4ever.com> wrote:
Where on your diagram is the Router? I assume that your Kamailio 1 and Kamailio 2 are on two different networks?
Paul
On Wed, 31 Jul, 2013 at 10:12 PM, Peter <eidevm5@gmail.com> wrote:
After checking everything for the umpteenth time, I discovered I was missing a route back to 10.1.254.0/24
Once I put that in, bingo, audio!
On Thu, Aug 1, 2013 at 12:33 PM, Peter <eidevm5@gmail.com> wrote:
I currently have 2 SIP clients (Linphone) successfully calling each other, but there is no audio on either end.
The set up is as follows:
Linphone1 (1000) --> Kamailio 1 <-------> Freeswitch <------> Kamalio 2 <---- Linphone2 (2000)
Using Freeswitch 1.2.12 on CentOS (installed via RPM)
Freeswitch has two interfaces:
external - 10.1.1.206 internal - 10.10.10.206
Each of the Linphone clients are registered their respective Kamailio instance and Kamailio is configured to route via the appropriate interface on Freeswitch.
The SIP negotiation is working as I can call either Linphone client.
I've done a tcpdump on each side of Freeswitch and can see the RTP traffic between the Linphone and the appropriate interface on Freeswitch.
I've tried different codec combinations (mostly G711 and iLBC), and different SIP clients but still get no audio.
Any pointers on how to track down the issue?
Thanks
Peter
_________________________________________________________________________ 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'm sorry, it looks like I missed the part where you mentioned that you have it all working now, that's why I was asking about where you router is because it very much sounded like routing/nat issues you were having :) Paul On Thu, 1 Aug, 2013 at 6:34 PM, Peter <eidevm5@gmail.com> wrote:
Correct. Both Kamailio servers are on different networks. I didn't show the router in the mix for simplicity.
On Fri, Aug 2, 2013 at 6:31 AM, Paul <pasha@prosperity4ever.com> wrote:
Where on your diagram is the Router? I assume that your Kamailio 1 and Kamailio 2 are on two different networks?
Paul
On Wed, 31 Jul, 2013 at 10:12 PM, Peter <eidevm5@gmail.com> wrote:
After checking everything for the umpteenth time, I discovered I was missing a route back to 10.1.254.0/24
Once I put that in, bingo, audio!
On Thu, Aug 1, 2013 at 12:33 PM, Peter <eidevm5@gmail.com> wrote:
I currently have 2 SIP clients (Linphone) successfully calling each other, but there is no audio on either end.
The set up is as follows:
Linphone1 (1000) --> Kamailio 1 <-------> Freeswitch <------> Kamalio 2 <---- Linphone2 (2000)
Using Freeswitch 1.2.12 on CentOS (installed via RPM)
Freeswitch has two interfaces:
external - 10.1.1.206 internal - 10.10.10.206
Each of the Linphone clients are registered their respective Kamailio instance and Kamailio is configured to route via the appropriate interface on Freeswitch.
The SIP negotiation is working as I can call either Linphone client.
I've done a tcpdump on each side of Freeswitch and can see the RTP traffic between the Linphone and the appropriate interface on Freeswitch.
I've tried different codec combinations (mostly G711 and iLBC), and different SIP clients but still get no audio.
Any pointers on how to track down the issue?
Thanks
Peter
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
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Michael Collins
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Paul
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Peter