Automatic transcoding does not seem to occur.
I've got a client with OPUS and G.711 codecs enabled and FreeSWITCH with OPUS and G.711 codecs enabled. I've got a SIP gateway connected to FreeSWITCH with only support for G.711. I want to have clients only use OPUS codec - as this seems best for mobile data connections. As of now client <-> FS <-> client use OPUS, but whenever the SIP gateway is involved G.711 is chosen. If I disable G.711 codec on the client side, to force OPUS, I'm not allowed to dial out via my SIP gateway. So transcoding to G.711 does not occur. My fs_cli reports this: Hangup sofia/external/+4413265874 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] Again, reenabling G.711 in the client and everything works. I have not set disable-transcoding=true. What to do?
My setup is based upon https://github.com/voxserv/freeswitch_conf_minimal. On Wed, Sep 16, 2015 at 1:26 PM, Michael Nielsen <mic.niel84@gmail.com> wrote:
I've got a client with OPUS and G.711 codecs enabled and FreeSWITCH with OPUS and G.711 codecs enabled. I've got a SIP gateway connected to FreeSWITCH with only support for G.711.
I want to have clients only use OPUS codec - as this seems best for mobile data connections.
As of now client <-> FS <-> client use OPUS, but whenever the SIP gateway is involved G.711 is chosen.
If I disable G.711 codec on the client side, to force OPUS, I'm not allowed to dial out via my SIP gateway. So transcoding to G.711 does not occur.
My fs_cli reports this: Hangup sofia/external/+4413265874 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
Again, reenabling G.711 in the client and everything works.
I have not set disable-transcoding=true.
What to do?
Hi Michael, Is your Opus codec loaded ? Do you see these in FS log ? 2015-09-16 11:27:00.235630 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 40ms (VBR) 2015-09-16 11:27:00.235644 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 40ms (VBR) 2015-09-16 11:27:00.235651 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 120ms (VBR) 2015-09-16 11:27:00.235658 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 100ms (VBR) 2015-09-16 11:27:00.235663 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 80ms (VBR) 2015-09-16 11:27:00.235669 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 60ms (VBR) 2015-09-16 11:27:00.235675 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 20ms (VBR) 2015-09-16 11:27:00.235680 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 20ms (VBR) 2015-09-16 11:27:00.235685 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 10ms (VBR) 2015-09-16 11:27:00.235690 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 10ms (VBR) 2015-09-16 11:27:00.235696 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 40ms (VBR) 2015-09-16 11:27:00.235701 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 20ms (VBR) 2015-09-16 11:27:00.235707 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 20ms (VBR) 2015-09-16 11:27:00.235712 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 10ms (VBR) 2015-09-16 11:27:00.235717 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 10ms (VBR) Any errors in your log ? Are u sure you bridge the call to the g.711 trunk ? I think your client-to-client calls work because is just doing media bypass or pass through . Regards, Dragos On 2015-09-16 13:31, Michael Nielsen wrote:
My setup is based upon https://github.com/voxserv/freeswitch_conf_minimal [1].
On Wed, Sep 16, 2015 at 1:26 PM, Michael Nielsen <mic.niel84@gmail.com> wrote:
I've got a client with OPUS and G.711 codecs enabled and FreeSWITCH with OPUS and G.711 codecs enabled. I've got a SIP gateway connected to FreeSWITCH with only support for G.711.
I want to have clients only use OPUS codec - as this seems best for mobile data connections.
As of now client <-> FS <-> client use OPUS, but whenever the SIP gateway is involved G.711 is chosen.
If I disable G.711 codec on the client side, to force OPUS, I'm not allowed to dial out via my SIP gateway. So transcoding to G.711 does not occur.
My fs_cli reports this: Hangup sofia/external/+4413265874 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
Again, reenabling G.711 in the client and everything works.
I have not set disable-transcoding=true.
What to do?
Links: ------ [1] https://github.com/voxserv/freeswitch_conf_minimal
_________________________________________________________________________ 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
We may need to revisit all your OPUS patches after the work that has been completed. On Wed, Sep 16, 2015 at 7:00 AM, doancea <doancea@orange-vallee.net> wrote:
Hi Michael,
Is your Opus codec loaded ? Do you see these in FS log ?
2015-09-16 11:27:00.235630 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 40ms (VBR) 2015-09-16 11:27:00.235644 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 40ms (VBR) 2015-09-16 11:27:00.235651 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 120ms (VBR) 2015-09-16 11:27:00.235658 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 100ms (VBR) 2015-09-16 11:27:00.235663 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 80ms (VBR) 2015-09-16 11:27:00.235669 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 60ms (VBR) 2015-09-16 11:27:00.235675 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 20ms (VBR) 2015-09-16 11:27:00.235680 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 20ms (VBR) 2015-09-16 11:27:00.235685 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 10ms (VBR) 2015-09-16 11:27:00.235690 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 10ms (VBR) 2015-09-16 11:27:00.235696 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 40ms (VBR) 2015-09-16 11:27:00.235701 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 20ms (VBR) 2015-09-16 11:27:00.235707 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 20ms (VBR) 2015-09-16 11:27:00.235712 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 10ms (VBR) 2015-09-16 11:27:00.235717 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 10ms (VBR)
Any errors in your log ? Are u sure you bridge the call to the g.711 trunk ? I think your client-to-client calls work because is just doing media bypass or pass through .
Regards, Dragos
On 2015-09-16 13:31, Michael Nielsen wrote:
My setup is based upon https://github.com/voxserv/freeswitch_conf_minimal [1].
On Wed, Sep 16, 2015 at 1:26 PM, Michael Nielsen <mic.niel84@gmail.com> wrote:
I've got a client with OPUS and G.711 codecs enabled and FreeSWITCH with OPUS and G.711 codecs enabled. I've got a SIP gateway connected to FreeSWITCH with only support for G.711.
I want to have clients only use OPUS codec - as this seems best for mobile data connections.
As of now client <-> FS <-> client use OPUS, but whenever the SIP gateway is involved G.711 is chosen.
If I disable G.711 codec on the client side, to force OPUS, I'm not allowed to dial out via my SIP gateway. So transcoding to G.711 does not occur.
My fs_cli reports this: Hangup sofia/external/+4413265874 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
Again, reenabling G.711 in the client and everything works.
I have not set disable-transcoding=true.
What to do?
Links: ------ [1] https://github.com/voxserv/freeswitch_conf_minimal
_________________________________________________________________________ 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
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Michael, Can you transcode with other codecs with your config ? If you don't see any output with "opus_debug on" while placing a call , it means you have a config issue. Dragos On 2015-09-16 15:17, Brian West wrote:
We may need to revisit all your OPUS patches after the work that has been completed.
On Wed, Sep 16, 2015 at 7:00 AM, doancea <doancea@orange-vallee.net> wrote:
Hi Michael,
Is your Opus codec loaded ? Do you see these in FS log ?
2015-09-16 11:27:00.235630 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 40ms (VBR) 2015-09-16 11:27:00.235644 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 40ms (VBR) 2015-09-16 11:27:00.235651 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 120ms (VBR) 2015-09-16 11:27:00.235658 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 100ms (VBR) 2015-09-16 11:27:00.235663 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 80ms (VBR) 2015-09-16 11:27:00.235669 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 60ms (VBR) 2015-09-16 11:27:00.235675 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 20ms (VBR) 2015-09-16 11:27:00.235680 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 20ms (VBR) 2015-09-16 11:27:00.235685 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 10ms (VBR) 2015-09-16 11:27:00.235690 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 8000hz 10ms (VBR) 2015-09-16 11:27:00.235696 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 40ms (VBR) 2015-09-16 11:27:00.235701 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 20ms (VBR) 2015-09-16 11:27:00.235707 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 20ms (VBR) 2015-09-16 11:27:00.235712 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 10ms (VBR) 2015-09-16 11:27:00.235717 [NOTICE] switch_loadable_module.c:213 Adding Codec opus 116 OPUS (STANDARD) 48000hz 10ms (VBR)
Any errors in your log ? Are u sure you bridge the call to the g.711 trunk ? I think your client-to-client calls work because is just doing media bypass or pass through .
Regards, Dragos
On 2015-09-16 13:31, Michael Nielsen wrote:
My setup is based upon https://github.com/voxserv/freeswitch_conf_minimal [1] [1].
On Wed, Sep 16, 2015 at 1:26 PM, Michael Nielsen <mic.niel84@gmail.com> wrote:
I've got a client with OPUS and G.711 codecs enabled and FreeSWITCH with OPUS and G.711 codecs enabled. I've got a SIP gateway connected to FreeSWITCH with only support for G.711.
I want to have clients only use OPUS codec - as this seems best for mobile data connections.
As of now client <-> FS <-> client use OPUS, but whenever the SIP gateway is involved G.711 is chosen.
If I disable G.711 codec on the client side, to force OPUS, I'm not allowed to dial out via my SIP gateway. So transcoding to G.711 does not occur.
My fs_cli reports this: Hangup sofia/external/+4413265874 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
Again, reenabling G.711 in the client and everything works.
I have not set disable-transcoding=true.
What to do?
Links: ------ [1] https://github.com/voxserv/freeswitch_conf_minimal [1]
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com [2]
Official FreeSWITCH Sites http://www.freeswitch.org [3] http://confluence.freeswitch.org [4] http://www.cluecon.com [5]
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[7]
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com [2]
Official FreeSWITCH Sites http://www.freeswitch.org [3] http://confluence.freeswitch.org [4] http://www.cluecon.com [5]
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--
_BRIAN WEST_ brian@freeswitch.org
_TWITTER: @FREESWITCH , @BRIANKWEST_ http://www.freeswitchbook.com [8] http://www.freeswitchcookbook.com [9]
Got Bugs? Report them here [10]! | Reddit: /r/freeswitch [11]
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) INUM:+883 5100 1420 9001 | ISN:410*543 | SKYPE:briankwest
Links: ------ [1] https://github.com/voxserv/freeswitch_conf_minimal [2] http://www.freeswitchsolutions.com [3] http://www.freeswitch.org [4] http://confluence.freeswitch.org [5] http://www.cluecon.com [6] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [7] http://lists.freeswitch.org/mailman/options/freeswitch-users [8] http://www.freeswitchbook.com [9] http://www.freeswitchcookbook.com [10] https://freeswitch.org/jira [11] https://www.reddit.com/r/freeswitch
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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participants (3)
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Brian West
-
doancea
-
Michael Nielsen