garbled audio with G726-32, other codecs are fine
Hello I'm migrating an office setup from asterisk to FS and in the process I was considering using G726-32 for some bandwidth starved remote endpoints. However I only get metallic/garbled audio with that codec even when simply playing moh to the endpoint, while other codecs work fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds marginally better but still garbled and really worse than G711. The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest (centos6 64bit host). But please don't shoot ! :) - I know about virtual environment limitations but for these tests the host is only lightly loaded, there aren't any calls to the FS instance except my tests, and the fact that it works with other codecs makes me think that virtualization is not the issue here. I may be wrong though. Is there any guide for debugging that kind of problem before reverting to a fresh install on bare-metal with the latest HEAD ? Until now I've tried: - improving timers ; but the default soft timer (which I guess uses timerd) works best. The time interval between sent packets on a tcpdump trace looks identical to the output of "timer_test", so that doesn't seem to be a network/jitter problem. And there's no problem with other codecs, but maybe G726-XX is specific. For info the guest's clocksource is kvm_clock, while the host uses tsc. - using different endpoints: the production ones are Linksys PAP2 ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the same thing happens with linphone on a fedora 19 laptop. A call with rtp media going through FS without transcoding - G726-32 to G726-32 - works perfectly (I can't hear the difference with G711). The problem is only when there's transcoding to G726 (from wav for moh, or from any other codec when bridging). I've looked at the wiki, posts, changelogs, jira, ..., but am a bit at a loss now. Any pointers ? Except that little problem, FS rocks, and I'm happy I can finally ditch asterisk. Kudos to the core devs and contributors. Ivan
AAC bitpacking by any chance? I thought I had a similar issue, happened so long ago I cant remember what I did. Thank you, Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com Indianapolis, Indiana Sent from a mobile device. On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev@c3i.bg> wrote:
Hello
I'm migrating an office setup from asterisk to FS and in the process I was considering using G726-32 for some bandwidth starved remote endpoints. However I only get metallic/garbled audio with that codec even when simply playing moh to the endpoint, while other codecs work fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds marginally better but still garbled and really worse than G711.
The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest (centos6 64bit host). But please don't shoot ! :) - I know about virtual environment limitations but for these tests the host is only lightly loaded, there aren't any calls to the FS instance except my tests, and the fact that it works with other codecs makes me think that virtualization is not the issue here. I may be wrong though.
Is there any guide for debugging that kind of problem before reverting to a fresh install on bare-metal with the latest HEAD ? Until now I've tried:
- improving timers ; but the default soft timer (which I guess uses timerd) works best. The time interval between sent packets on a tcpdump trace looks identical to the output of "timer_test", so that doesn't seem to be a network/jitter problem. And there's no problem with other codecs, but maybe G726-XX is specific. For info the guest's clocksource is kvm_clock, while the host uses tsc.
- using different endpoints: the production ones are Linksys PAP2 ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the same thing happens with linphone on a fedora 19 laptop.
A call with rtp media going through FS without transcoding - G726-32 to G726-32 - works perfectly (I can't hear the difference with G711). The problem is only when there's transcoding to G726 (from wav for moh, or from any other codec when bridging). I've looked at the wiki, posts, changelogs, jira, ..., but am a bit at a loss now.
Any pointers ?
Except that little problem, FS rocks, and I'm happy I can finally ditch asterisk. Kudos to the core devs and contributors.
Ivan
_________________________________________________________________________ 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
You can turn off G726 AAC bit-packing in spandsp.conf.xml. By the way, there are other codecs out there you can try. SPEEX comes to mind if all your endpoints don't deal with the PSTN. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian Foster Sent: Saturday, August 3, 2013 2:37 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] garbled audio with G726-32, other codecs are fine AAC bitpacking by any chance? I thought I had a similar issue, happened so long ago I cant remember what I did. Thank you, Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com Indianapolis, Indiana Sent from a mobile device. On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev@c3i.bg> wrote: Hello I'm migrating an office setup from asterisk to FS and in the process I was considering using G726-32 for some bandwidth starved remote endpoints. However I only get metallic/garbled audio with that codec even when simply playing moh to the endpoint, while other codecs work fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds marginally better but still garbled and really worse than G711. The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest (centos6 64bit host). But please don't shoot ! :) - I know about virtual environment limitations but for these tests the host is only lightly loaded, there aren't any calls to the FS instance except my tests, and the fact that it works with other codecs makes me think that virtualization is not the issue here. I may be wrong though. Is there any guide for debugging that kind of problem before reverting to a fresh install on bare-metal with the latest HEAD ? Until now I've tried: - improving timers ; but the default soft timer (which I guess uses timerd) works best. The time interval between sent packets on a tcpdump trace looks identical to the output of "timer_test", so that doesn't seem to be a network/jitter problem. And there's no problem with other codecs, but maybe G726-XX is specific. For info the guest's clocksource is kvm_clock, while the host uses tsc. - using different endpoints: the production ones are Linksys PAP2 ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the same thing happens with linphone on a fedora 19 laptop. A call with rtp media going through FS without transcoding - G726-32 to G726-32 - works perfectly (I can't hear the difference with G711). The problem is only when there's transcoding to G726 (from wav for moh, or from any other codec when bridging). I've looked at the wiki, posts, changelogs, jira, ..., but am a bit at a loss now. Any pointers ? Except that little problem, FS rocks, and I'm happy I can finally ditch asterisk. Kudos to the core devs and contributors. Ivan _________________________________________________________________________ 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
Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to the linphone client I only got cracks and whitenoise, I've forgot to mention that in my post. That said I've tried to uncomment and set <param name="bitpacking" value="none"/> in internal.xml ("none" is a wild guess - I couldn't find any doc on values accepted by this parameter), but that doesn't help. Speex: the ATAs don't support it. And being stubborn I'd like to understand what's wrong with G726 :) On 08/04/2013 05:22 AM, Jeff Leung wrote:
You can turn off G726 AAC bit-packing in spandsp.conf.xml.
By the way, there are other codecs out there you can try. SPEEX comes to mind if all your endpoints don’t deal with the PSTN.
*From:*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian Foster *Sent:* Saturday, August 3, 2013 2:37 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other codecs are fine
AAC bitpacking by any chance? I thought I had a similar issue, happened so long ago I cant remember what I did.
Thank you,
Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com <mailto:bdfoster@davri.com> Indianapolis, Indiana
Sent from a mobile device.
On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev@c3i.bg <mailto:imitev@c3i.bg>> wrote:
Hello
I'm migrating an office setup from asterisk to FS and in the process I was considering using G726-32 for some bandwidth starved remote endpoints. However I only get metallic/garbled audio with that codec even when simply playing moh to the endpoint, while other codecs work fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds marginally better but still garbled and really worse than G711.
The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest (centos6 64bit host). But please don't shoot ! :) - I know about virtual environment limitations but for these tests the host is only lightly loaded, there aren't any calls to the FS instance except my tests, and the fact that it works with other codecs makes me think that virtualization is not the issue here. I may be wrong though.
Is there any guide for debugging that kind of problem before reverting to a fresh install on bare-metal with the latest HEAD ? Until now I've tried:
- improving timers ; but the default soft timer (which I guess uses timerd) works best. The time interval between sent packets on a tcpdump trace looks identical to the output of "timer_test", so that doesn't seem to be a network/jitter problem. And there's no problem with other codecs, but maybe G726-XX is specific. For info the guest's clocksource is kvm_clock, while the host uses tsc.
- using different endpoints: the production ones are Linksys PAP2 ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the same thing happens with linphone on a fedora 19 laptop.
A call with rtp media going through FS without transcoding - G726-32 to G726-32 - works perfectly (I can't hear the difference with G711). The problem is only when there's transcoding to G726 (from wav for moh, or from any other codec when bridging). I've looked at the wiki, posts, changelogs, jira, ..., but am a bit at a loss now.
Any pointers ?
Except that little problem, FS rocks, and I'm happy I can finally ditch asterisk. Kudos to the core devs and contributors.
Ivan
_________________________________________________________________________ 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
_________________________________________________________________________ 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
There is always bit packing, but there are 2 different ways to do the bit packing. A lot of devices get it wrong so its worth looking at that. Mike On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev@c3i.bg> wrote:
Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to the linphone client I only got cracks and whitenoise, I've forgot to mention that in my post.
That said I've tried to uncomment and set <param name="bitpacking" value="none"/> in internal.xml ("none" is a wild guess - I couldn't find any doc on values accepted by this parameter), but that doesn't help.
Speex: the ATAs don't support it. And being stubborn I'd like to understand what's wrong with G726 :)
On 08/04/2013 05:22 AM, Jeff Leung wrote:
You can turn off G726 AAC bit-packing in spandsp.conf.xml.
By the way, there are other codecs out there you can try. SPEEX comes to mind if all your endpoints don’t deal with the PSTN.
*From:*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian Foster *Sent:* Saturday, August 3, 2013 2:37 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other codecs are fine
AAC bitpacking by any chance? I thought I had a similar issue, happened so long ago I cant remember what I did.
Thank you,
Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com <mailto:bdfoster@davri.com> Indianapolis, Indiana
Sent from a mobile device.
On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev@c3i.bg <mailto:imitev@c3i.bg>> wrote:
Hello
I'm migrating an office setup from asterisk to FS and in the process I was considering using G726-32 for some bandwidth starved remote endpoints. However I only get metallic/garbled audio with that codec even when simply playing moh to the endpoint, while other codecs work fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds marginally better but still garbled and really worse than G711.
The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest (centos6 64bit host). But please don't shoot ! :) - I know about virtual environment limitations but for these tests the host is only lightly loaded, there aren't any calls to the FS instance except my tests, and the fact that it works with other codecs makes me think that virtualization is not the issue here. I may be wrong though.
Is there any guide for debugging that kind of problem before reverting to a fresh install on bare-metal with the latest HEAD ? Until now I've tried:
- improving timers ; but the default soft timer (which I guess uses timerd) works best. The time interval between sent packets on a tcpdump trace looks identical to the output of "timer_test", so that doesn't seem to be a network/jitter problem. And there's no problem with other codecs, but maybe G726-XX is specific. For info the guest's clocksource is kvm_clock, while the host uses tsc.
- using different endpoints: the production ones are Linksys PAP2 ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the same thing happens with linphone on a fedora 19 laptop.
A call with rtp media going through FS without transcoding - G726-32 to G726-32 - works perfectly (I can't hear the difference with G711). The problem is only when there's transcoding to G726 (from wav for moh, or from any other codec when bridging). I've looked at the wiki, posts, changelogs, jira, ..., but am a bit at a loss now.
Any pointers ?
Except that little problem, FS rocks, and I'm happy I can finally ditch asterisk. Kudos to the core devs and contributors.
Ivan
Ah, I didn't know there was always bit packing - I guess I should read how G726 works, thanks for pointing that. The thing is, both linksys PAPs and linphone have the same problem, and they are not "exotic" endpoints so that makes me think the cause is my freeswitch setup. I'll try to test with Xlite when I have a chance, and also see if wireshark shows enough codec detail to find out if the endpoints get it wrong. I'm of course interested if you have other ideas on how to debug that. ivan On 08/12/2013 05:14 PM, Michael Jerris wrote:
There is always bit packing, but there are 2 different ways to do the bit packing. A lot of devices get it wrong so its worth looking at that.
Mike
On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev@c3i.bg> wrote:
Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to the linphone client I only got cracks and whitenoise, I've forgot to mention that in my post.
That said I've tried to uncomment and set <param name="bitpacking" value="none"/> in internal.xml ("none" is a wild guess - I couldn't find any doc on values accepted by this parameter), but that doesn't help.
Speex: the ATAs don't support it. And being stubborn I'd like to understand what's wrong with G726 :)
On 08/04/2013 05:22 AM, Jeff Leung wrote:
You can turn off G726 AAC bit-packing in spandsp.conf.xml.
By the way, there are other codecs out there you can try. SPEEX comes to mind if all your endpoints don’t deal with the PSTN.
*From:*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian Foster *Sent:* Saturday, August 3, 2013 2:37 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other codecs are fine
AAC bitpacking by any chance? I thought I had a similar issue, happened so long ago I cant remember what I did.
Thank you,
Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com <mailto:bdfoster@davri.com> Indianapolis, Indiana
Sent from a mobile device.
On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev@c3i.bg <mailto:imitev@c3i.bg>> wrote:
Hello
I'm migrating an office setup from asterisk to FS and in the process I was considering using G726-32 for some bandwidth starved remote endpoints. However I only get metallic/garbled audio with that codec even when simply playing moh to the endpoint, while other codecs work fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds marginally better but still garbled and really worse than G711.
The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest (centos6 64bit host). But please don't shoot ! :) - I know about virtual environment limitations but for these tests the host is only lightly loaded, there aren't any calls to the FS instance except my tests, and the fact that it works with other codecs makes me think that virtualization is not the issue here. I may be wrong though.
Is there any guide for debugging that kind of problem before reverting to a fresh install on bare-metal with the latest HEAD ? Until now I've tried:
- improving timers ; but the default soft timer (which I guess uses timerd) works best. The time interval between sent packets on a tcpdump trace looks identical to the output of "timer_test", so that doesn't seem to be a network/jitter problem. And there's no problem with other codecs, but maybe G726-XX is specific. For info the guest's clocksource is kvm_clock, while the host uses tsc.
- using different endpoints: the production ones are Linksys PAP2 ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the same thing happens with linphone on a fedora 19 laptop.
A call with rtp media going through FS without transcoding - G726-32 to G726-32 - works perfectly (I can't hear the difference with G711). The problem is only when there's transcoding to G726 (from wav for moh, or from any other codec when bridging). I've looked at the wiki, posts, changelogs, jira, ..., but am a bit at a loss now.
Any pointers ?
Except that little problem, FS rocks, and I'm happy I can finally ditch asterisk. Kudos to the core devs and contributors.
Ivan
_________________________________________________________________________ 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
Just found some time to do more tests: in the end it was simply a matter of enabling aal2 bitpacking in sofia. I really didn't think that would solve the problem since both endpoints had the same problem. (BTW, looking back at the interop page with the right set of keywords, I now see that someone had the same problem with a linksys SPA2000). Anyway, thanks to everybody who replied with pointers to bitpacking - I assumed wrongly there was no bitpacking at all on G726. Note: as I'm not knowledgeable enough on G726/bitpacking to update the wiki, I've set up a bounty - see the wiki page. On 08/13/2013 09:57 AM, Ivan wrote:
Ah, I didn't know there was always bit packing - I guess I should read how G726 works, thanks for pointing that.
The thing is, both linksys PAPs and linphone have the same problem, and they are not "exotic" endpoints so that makes me think the cause is my freeswitch setup. I'll try to test with Xlite when I have a chance, and also see if wireshark shows enough codec detail to find out if the endpoints get it wrong. I'm of course interested if you have other ideas on how to debug that.
ivan
On 08/12/2013 05:14 PM, Michael Jerris wrote:
There is always bit packing, but there are 2 different ways to do the bit packing. A lot of devices get it wrong so its worth looking at that.
Mike
On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev@c3i.bg> wrote:
Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to the linphone client I only got cracks and whitenoise, I've forgot to mention that in my post.
That said I've tried to uncomment and set <param name="bitpacking" value="none"/> in internal.xml ("none" is a wild guess - I couldn't find any doc on values accepted by this parameter), but that doesn't help.
Speex: the ATAs don't support it. And being stubborn I'd like to understand what's wrong with G726 :)
On 08/04/2013 05:22 AM, Jeff Leung wrote:
You can turn off G726 AAC bit-packing in spandsp.conf.xml.
By the way, there are other codecs out there you can try. SPEEX comes to mind if all your endpoints don’t deal with the PSTN.
*From:*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian Foster *Sent:* Saturday, August 3, 2013 2:37 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other codecs are fine
AAC bitpacking by any chance? I thought I had a similar issue, happened so long ago I cant remember what I did.
Thank you,
Brian Foster Project Manager/Owner's Rep. Davri Investments, Inc. O: 317-787-2686 x2102 M: 317-600-9753 E: bdfoster@davri.com <mailto:bdfoster@davri.com> Indianapolis, Indiana
Sent from a mobile device.
On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev@c3i.bg <mailto:imitev@c3i.bg>> wrote:
Hello
I'm migrating an office setup from asterisk to FS and in the process I was considering using G726-32 for some bandwidth starved remote endpoints. However I only get metallic/garbled audio with that codec even when simply playing moh to the endpoint, while other codecs work fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds marginally better but still garbled and really worse than G711.
The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest (centos6 64bit host). But please don't shoot ! :) - I know about virtual environment limitations but for these tests the host is only lightly loaded, there aren't any calls to the FS instance except my tests, and the fact that it works with other codecs makes me think that virtualization is not the issue here. I may be wrong though.
Is there any guide for debugging that kind of problem before reverting to a fresh install on bare-metal with the latest HEAD ? Until now I've tried:
- improving timers ; but the default soft timer (which I guess uses timerd) works best. The time interval between sent packets on a tcpdump trace looks identical to the output of "timer_test", so that doesn't seem to be a network/jitter problem. And there's no problem with other codecs, but maybe G726-XX is specific. For info the guest's clocksource is kvm_clock, while the host uses tsc.
- using different endpoints: the production ones are Linksys PAP2 ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the same thing happens with linphone on a fedora 19 laptop.
A call with rtp media going through FS without transcoding - G726-32 to G726-32 - works perfectly (I can't hear the difference with G711). The problem is only when there's transcoding to G726 (from wav for moh, or from any other codec when bridging). I've looked at the wiki, posts, changelogs, jira, ..., but am a bit at a loss now.
Any pointers ?
Except that little problem, FS rocks, and I'm happy I can finally ditch asterisk. Kudos to the core devs and contributors.
Ivan
_________________________________________________________________________ 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
_________________________________________________________________________ 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 (5)
-
Brian Foster
-
Ivan
-
Ivan Mitev
-
Jeff Leung
-
Michael Jerris