L16 Codec in mod_conference
Hey all, I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes: session:execute("pre_answer") session:execute("conference_set_auto_outcall", "user/" .. args.call_user) session:execute("conference", "testroom@default") My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed: variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: opus But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed: variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: L16 Both tests are based on Linphone offical app. During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. Any suggestion would be appreciated. Regards, C.Wang
What rate is your conference running at?
On Nov 12, 2019, at 6:49 PM, 王聡 <cong.wang.itsherpa@gmail.com> wrote:
Hey all,
I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes:
session:execute("pre_answer") session:execute("conference_set_auto_outcall", "user/" .. args.call_user) session:execute("conference", "testroom@default")
My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: opus
But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: L16
Both tests are based on Linphone offical app. During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. Any suggestion would be appreciated.
Perhaps 48000, from uuid_dump: variable_rtp_use_codec_rate: 48000 Regards.
在 2019年11月14日，04:25，Mike Jerris <mike@freeswitch.org> 写道：
What rate is your conference running at?
On Nov 12, 2019, at 6:49 PM, 王聡 <cong.wang.itsherpa@gmail.com <mailto:cong.wang.itsherpa@gmail.com>> wrote:
Hey all,
I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes:
session:execute("pre_answer") session:execute("conference_set_auto_outcall", "user/" .. args.call_user) session:execute("conference", "testroom@default")
My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: opus
But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: L16
Both tests are based on Linphone offical app. During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. Any suggestion would be appreciated.
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
Professional FreeSWITCH Services sales@freeswitch.com <mailto:sales@freeswitch.com> https://freeswitch.com <https://freeswitch.com/>
Official FreeSWITCH Sites https://freeswitch.com/oss <https://freeswitch.com/oss> https://freeswitch.org/confluence <https://freeswitch.org/confluence> https://cluecon.com <https://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> https://freeswitch.com <https://freeswitch.com/>
Your issue does not have anything to do with L16 per se, and you cannot disable it for mod_conference. Everything is decoded into L16 / encoded from L16, like all audio codecs do. A conference is more CPU intensive than a bridge. On Thu, Nov 14, 2019 at 9:43 AM 王聡 <cong.wang.itsherpa@gmail.com> wrote:
Perhaps 48000, from uuid_dump:
variable_rtp_use_codec_rate: 48000
Regards.
在 2019年11月14日，04:25，Mike Jerris <mike@freeswitch.org> 写道：
What rate is your conference running at?
On Nov 12, 2019, at 6:49 PM, 王聡 <cong.wang.itsherpa@gmail.com> wrote:
Hey all,
I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes:
session:execute("pre_answer") session:execute("conference_set_auto_outcall", "user/" .. args.call_user) session:execute("conference", "testroom@default")
My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: opus
But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: L16
Both tests are based on Linphone offical app. During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. Any suggestion would be appreciated.
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
Thanks for your reply. However we had found the audio quality under conference mod is not as good as bridge mode, though both OPUS. I wonder if this issue has any relations on L16 or not. Regards.
在 2019年11月15日，00:54，Dragos Oancea <dragos@freeswitch.org> 写道：
Your issue does not have anything to do with L16 per se, and you cannot disable it for mod_conference. Everything is decoded into L16 / encoded from L16, like all audio codecs do. A conference is more CPU intensive than a bridge.
On Thu, Nov 14, 2019 at 9:43 AM 王聡 <cong.wang.itsherpa@gmail.com <mailto:cong.wang.itsherpa@gmail.com>> wrote: Perhaps 48000, from uuid_dump:
variable_rtp_use_codec_rate: 48000
Regards.
在 2019年11月14日，04:25，Mike Jerris <mike@freeswitch.org <mailto:mike@freeswitch.org>> 写道：
What rate is your conference running at?
On Nov 12, 2019, at 6:49 PM, 王聡 <cong.wang.itsherpa@gmail.com <mailto:cong.wang.itsherpa@gmail.com>> wrote:
Hey all,
I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes:
session:execute("pre_answer") session:execute("conference_set_auto_outcall", "user/" .. args.call_user) session:execute("conference", "testroom@default")
My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: opus
But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: L16
Both tests are based on Linphone offical app. During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. Any suggestion would be appreciated.
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
Professional FreeSWITCH Services sales@freeswitch.com <mailto:sales@freeswitch.com> https://freeswitch.com <https://freeswitch.com/>
Official FreeSWITCH Sites https://freeswitch.com/oss <https://freeswitch.com/oss> https://freeswitch.org/confluence <https://freeswitch.org/confluence> https://cluecon.com <https://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> https://freeswitch.com <https://freeswitch.com/>
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
Professional FreeSWITCH Services sales@freeswitch.com <mailto:sales@freeswitch.com> https://freeswitch.com <https://freeswitch.com/>
Official FreeSWITCH Sites https://freeswitch.com/oss <https://freeswitch.com/oss> https://freeswitch.org/confluence <https://freeswitch.org/confluence> https://cluecon.com <https://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> https://freeswitch.com <https://freeswitch.com/>_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
No, thats the codec rate. This will be in your conference configuration.
On Nov 14, 2019, at 1:42 AM, 王聡 <cong.wang.itsherpa@gmail.com> wrote:
Perhaps 48000, from uuid_dump:
variable_rtp_use_codec_rate: 48000
Regards.
在 2019年11月14日，04:25，Mike Jerris <mike@freeswitch.org <mailto:mike@freeswitch.org>> 写道：
What rate is your conference running at?
On Nov 12, 2019, at 6:49 PM, 王聡 <cong.wang.itsherpa@gmail.com <mailto:cong.wang.itsherpa@gmail.com>> wrote:
Hey all,
I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes:
session:execute("pre_answer") session:execute("conference_set_auto_outcall", "user/" .. args.call_user) session:execute("conference", "testroom@default")
My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: opus
But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: L16
Both tests are based on Linphone offical app. During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. Any suggestion would be appreciated.
We hadn’t configure this part, perhaps default? Sorry I’m newcomer to mod_conference. Regards.
在 2019年11月15日，03:26，Mike Jerris <mike@freeswitch.org> 写道：
No, thats the codec rate. This will be in your conference configuration.
On Nov 14, 2019, at 1:42 AM, 王聡 <cong.wang.itsherpa@gmail.com <mailto:cong.wang.itsherpa@gmail.com>> wrote:
Perhaps 48000, from uuid_dump:
variable_rtp_use_codec_rate: 48000
Regards.
在 2019年11月14日，04:25，Mike Jerris <mike@freeswitch.org <mailto:mike@freeswitch.org>> 写道：
What rate is your conference running at?
On Nov 12, 2019, at 6:49 PM, 王聡 <cong.wang.itsherpa@gmail.com <mailto:cong.wang.itsherpa@gmail.com>> wrote:
Hey all,
I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes:
session:execute("pre_answer") session:execute("conference_set_auto_outcall", "user/" .. args.call_user) session:execute("conference", "testroom@default")
My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: opus
But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed:
variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: L16
Both tests are based on Linphone offical app. During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. Any suggestion would be appreciated.
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
participants (3)
-
Dragos Oancea
-
Mike Jerris
-
王聡