Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny? We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711. How much processing power would it take to transcode from OPUS to G711? How well is the OPUS codec implemented on newer IP phones? I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it? Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram.
30% of packet loss ... are you mad? ... there is no way any codec could recover from that. Most of codecs, you will notice whenever you have more than 1-3% packet loss ... 30% ? ... really? ... should sound like and old synthetizer 2017-05-01 20:03 GMT+01:00 Colton Conor <colton.conor@gmail.com>:
Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny?
We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711.
How much processing power would it take to transcode from OPUS to G711?
How well is the OPUS codec implemented on newer IP phones?
I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it?
Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Raul, I'm sure the OP was referring to 30% packet loss with FEC, which opus can turn back into a usable audio stream. You'll absolutely notice artifacts in the audio, and you'd clearly identify something is wrong, but it's a heck of a lot better than 30% packet loss without FEC. Colton, Freeswitch supports opus, yes. I'd be curious to know why you want to transcode between opus and G.711, unless your endpoints don't support G.711 though. Yes, FEC can help you with packet loss on one end, but if your packet loss is on the G.711 end, then you're SOL anyway, so it seems pointless to do this unless you can use opus all the way though. As to how it will sound if doing the conversion, well, you're transcoding from one lossy codec to another lossy codec. The end result will be audio that will not sound as good as if it were just encoded in G.711 to begin with. Remember, the nature of lossy codecs is that they're, well, lossy. You can't make a G.711 stream sound better by re-encoding it to opus. Once it's encoded in G.711, that information is lost. As for CPU usage, opus is not particularly cheap. Transcoding + resampling (if required), will considerably limit the number of concurrent calls your FS instance can handle. Actual results are going to depend on a number of factors that make it infeasible to talk about it. But again, you probably don't want to be transcoding between these two codecs unless you absolutely have two (i.e. you can't get the two endpoints to otherwise agree on a codec). You're not going to gain any real advantage from doing this. Best, Colin On Mon, May 1, 2017 at 3:54 PM, Raúl Alexis Betancor Santana < rbetancor@gmail.com> wrote:
30% of packet loss ... are you mad? ... there is no way any codec could recover from that. Most of codecs, you will notice whenever you have more than 1-3% packet loss ... 30% ? ... really? ... should sound like and old synthetizer
2017-05-01 20:03 GMT+01:00 Colton Conor <colton.conor@gmail.com>:
Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny?
We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711.
How much processing power would it take to transcode from OPUS to G711?
How well is the OPUS codec implemented on newer IP phones?
I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it?
Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram.
_________________________________________________________________________ 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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I disagree… plenty of situations where you have a known quality leg to a provider over g711, and a uncontrolled internet leg to customer that you want to use opus/fec for.
On May 1, 2017, at 4:07 PM, Colin Morelli <colin.morelli@gmail.com> wrote:
Raul, I'm sure the OP was referring to 30% packet loss with FEC, which opus can turn back into a usable audio stream. You'll absolutely notice artifacts in the audio, and you'd clearly identify something is wrong, but it's a heck of a lot better than 30% packet loss without FEC.
Colton,
Freeswitch supports opus, yes. I'd be curious to know why you want to transcode between opus and G.711, unless your endpoints don't support G.711 though. Yes, FEC can help you with packet loss on one end, but if your packet loss is on the G.711 end, then you're SOL anyway, so it seems pointless to do this unless you can use opus all the way though.
As to how it will sound if doing the conversion, well, you're transcoding from one lossy codec to another lossy codec. The end result will be audio that will not sound as good as if it were just encoded in G.711 to begin with. Remember, the nature of lossy codecs is that they're, well, lossy. You can't make a G.711 stream sound better by re-encoding it to opus. Once it's encoded in G.711, that information is lost.
As for CPU usage, opus is not particularly cheap. Transcoding + resampling (if required), will considerably limit the number of concurrent calls your FS instance can handle. Actual results are going to depend on a number of factors that make it infeasible to talk about it. But again, you probably don't want to be transcoding between these two codecs unless you absolutely have two (i.e. you can't get the two endpoints to otherwise agree on a codec). You're not going to gain any real advantage from doing this.
Best, Colin
On Mon, May 1, 2017 at 3:54 PM, Raúl Alexis Betancor Santana <rbetancor@gmail.com <mailto:rbetancor@gmail.com>> wrote: 30% of packet loss ... are you mad? ... there is no way any codec could recover from that. Most of codecs, you will notice whenever you have more than 1-3% packet loss ... 30% ? ... really? ... should sound like and old synthetizer
2017-05-01 20:03 GMT+01:00 Colton Conor <colton.conor@gmail.com <mailto:colton.conor@gmail.com>>: Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny?
We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711.
How much processing power would it take to transcode from OPUS to G711?
How well is the OPUS codec implemented on newer IP phones?
I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it?
Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram.
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That's fair, if the primary focus of using the codec is purely to overcome a packet loss issue; however, consistent 30% packet loss to your endpoints seems like a fairly extreme scenario. I suppose if you anticipate a high enough level of packet loss on the average call to justify the slightly degraded audio quality that will be introduced by transcoding, then it may be worthwhile. I'll stand by the rest of my points though! Best, Colin On Mon, May 1, 2017 at 4:13 PM, Michael Jerris <mike@jerris.com> wrote:
I disagree… plenty of situations where you have a known quality leg to a provider over g711, and a uncontrolled internet leg to customer that you want to use opus/fec for.
On May 1, 2017, at 4:07 PM, Colin Morelli <colin.morelli@gmail.com> wrote:
Raul, I'm sure the OP was referring to 30% packet loss with FEC, which opus can turn back into a usable audio stream. You'll absolutely notice artifacts in the audio, and you'd clearly identify something is wrong, but it's a heck of a lot better than 30% packet loss without FEC.
Colton,
Freeswitch supports opus, yes. I'd be curious to know why you want to transcode between opus and G.711, unless your endpoints don't support G.711 though. Yes, FEC can help you with packet loss on one end, but if your packet loss is on the G.711 end, then you're SOL anyway, so it seems pointless to do this unless you can use opus all the way though.
As to how it will sound if doing the conversion, well, you're transcoding from one lossy codec to another lossy codec. The end result will be audio that will not sound as good as if it were just encoded in G.711 to begin with. Remember, the nature of lossy codecs is that they're, well, lossy. You can't make a G.711 stream sound better by re-encoding it to opus. Once it's encoded in G.711, that information is lost.
As for CPU usage, opus is not particularly cheap. Transcoding + resampling (if required), will considerably limit the number of concurrent calls your FS instance can handle. Actual results are going to depend on a number of factors that make it infeasible to talk about it. But again, you probably don't want to be transcoding between these two codecs unless you absolutely have two (i.e. you can't get the two endpoints to otherwise agree on a codec). You're not going to gain any real advantage from doing this.
Best, Colin
On Mon, May 1, 2017 at 3:54 PM, Raúl Alexis Betancor Santana < rbetancor@gmail.com> wrote:
30% of packet loss ... are you mad? ... there is no way any codec could recover from that. Most of codecs, you will notice whenever you have more than 1-3% packet loss ... 30% ? ... really? ... should sound like and old synthetizer
2017-05-01 20:03 GMT+01:00 Colton Conor <colton.conor@gmail.com>:
Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny?
We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711.
How much processing power would it take to transcode from OPUS to G711?
How well is the OPUS codec implemented on newer IP phones?
I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it?
Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram.
____________________________________________________________ _____________ 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
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I am just judging the 30 percent packet loss with FEC. Example are on the OPUS website https://opus-codec.org/examples/ Sounds pretty good to me. I would be terrified to know what G711 sounds like without FEC having evien 10 percent packet loss. I think there are many cases where one leg of the audio is uncontrolled, and the other leg of the audio via G711 is stable. For example, a freeswitch server sitting in a data center with redundant gigabit internet connections connected to a origination/termination carrier via G711. Then clients using OTT internet connections like their home cable connection, LTE, or wifi. The clients are the leg I want in OPUS to communicate back to the freeswitch server. Yes, the clients are able to speak G711 too, but I want OPUS for the FEC and bandwidth control. Are you saying transcoding from OPUS client to G711 is going to sound worse than just talking G711 all the way through? So what can a new Dell server with most recent gen intel xeon processors handle as far as transcoding goes? On Mon, May 1, 2017 at 3:19 PM, Colin Morelli <colin.morelli@gmail.com> wrote:
That's fair, if the primary focus of using the codec is purely to overcome a packet loss issue; however, consistent 30% packet loss to your endpoints seems like a fairly extreme scenario.
I suppose if you anticipate a high enough level of packet loss on the average call to justify the slightly degraded audio quality that will be introduced by transcoding, then it may be worthwhile.
I'll stand by the rest of my points though!
Best, Colin
On Mon, May 1, 2017 at 4:13 PM, Michael Jerris <mike@jerris.com> wrote:
I disagree… plenty of situations where you have a known quality leg to a provider over g711, and a uncontrolled internet leg to customer that you want to use opus/fec for.
On May 1, 2017, at 4:07 PM, Colin Morelli <colin.morelli@gmail.com> wrote:
Raul, I'm sure the OP was referring to 30% packet loss with FEC, which opus can turn back into a usable audio stream. You'll absolutely notice artifacts in the audio, and you'd clearly identify something is wrong, but it's a heck of a lot better than 30% packet loss without FEC.
Colton,
Freeswitch supports opus, yes. I'd be curious to know why you want to transcode between opus and G.711, unless your endpoints don't support G.711 though. Yes, FEC can help you with packet loss on one end, but if your packet loss is on the G.711 end, then you're SOL anyway, so it seems pointless to do this unless you can use opus all the way though.
As to how it will sound if doing the conversion, well, you're transcoding from one lossy codec to another lossy codec. The end result will be audio that will not sound as good as if it were just encoded in G.711 to begin with. Remember, the nature of lossy codecs is that they're, well, lossy. You can't make a G.711 stream sound better by re-encoding it to opus. Once it's encoded in G.711, that information is lost.
As for CPU usage, opus is not particularly cheap. Transcoding + resampling (if required), will considerably limit the number of concurrent calls your FS instance can handle. Actual results are going to depend on a number of factors that make it infeasible to talk about it. But again, you probably don't want to be transcoding between these two codecs unless you absolutely have two (i.e. you can't get the two endpoints to otherwise agree on a codec). You're not going to gain any real advantage from doing this.
Best, Colin
On Mon, May 1, 2017 at 3:54 PM, Raúl Alexis Betancor Santana < rbetancor@gmail.com> wrote:
30% of packet loss ... are you mad? ... there is no way any codec could recover from that. Most of codecs, you will notice whenever you have more than 1-3% packet loss ... 30% ? ... really? ... should sound like and old synthetizer
2017-05-01 20:03 GMT+01:00 Colton Conor <colton.conor@gmail.com>:
Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny?
We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711.
How much processing power would it take to transcode from OPUS to G711?
How well is the OPUS codec implemented on newer IP phones?
I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it?
Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram.
____________________________________________________________ _____________ 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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I agree, and was wrong before to suggest that there's no real advantage. FEC can be helpful to combat packet loss on unpredictable networks. Although again, I'll stand by my statement that consistent 10% packet loss would be pretty significant on any modern internet connection. Even consistent 2% packet loss would be cause for some concern. Intermittent packet loss issues, sure; but sustained 10% packet loss would be fairly alarming to hear about on any large scale. At that level, it's likely the user has problems with a number of services (even TCP ones, as they're likely to be experiencing slower speeds). On a consumer internet connection, your more frequent issues are almost certainly going to be latency and jitter, and generally your first line of defense (at least for jitter) would be the endpoints' jitter buffers. Although FEC can also help if you have to start dropping packets from the jitter buffer. Again, I'm not trying to claim it can't or won't happen. It can, and opus will help when it does. I'm just saying it probably makes sense to actually confirm this is the issue (if you're currently experiencing problems). Maybe my experiences have been different. *Are you saying transcoding from OPUS client to G711 is going to sound worse than just talking G711 all the way through? * Yes. Both codecs are lossy. Any time you encode to a lossy codec, you lose data. In this case, you're doing it twice. The originating device will be encoding raw audio to G.711, and FS will decode the G.711 and re-encode as opus. The degradation on the transcoding won't be huge, and it many cases it may not even really be noticeable (you'd have to test this in your own setup to see), but it will exist. I won't try to answer the question of how many transcoding channels any particular server can handle though. Perhaps the FS guys will have a decent answer, but I think they'll probably give similar feedback - you just need to test and find out. There are so many factors that come into play with performance that it becomes painfully difficult to try to guess. Best, Colin On Mon, May 1, 2017 at 5:21 PM, Colton Conor <colton.conor@gmail.com> wrote:
I am just judging the 30 percent packet loss with FEC. Example are on the OPUS website https://opus-codec.org/examples/ Sounds pretty good to me. I would be terrified to know what G711 sounds like without FEC having evien 10 percent packet loss.
I think there are many cases where one leg of the audio is uncontrolled, and the other leg of the audio via G711 is stable. For example, a freeswitch server sitting in a data center with redundant gigabit internet connections connected to a origination/termination carrier via G711.
Then clients using OTT internet connections like their home cable connection, LTE, or wifi. The clients are the leg I want in OPUS to communicate back to the freeswitch server. Yes, the clients are able to speak G711 too, but I want OPUS for the FEC and bandwidth control.
Are you saying transcoding from OPUS client to G711 is going to sound worse than just talking G711 all the way through?
So what can a new Dell server with most recent gen intel xeon processors handle as far as transcoding goes?
On Mon, May 1, 2017 at 3:19 PM, Colin Morelli <colin.morelli@gmail.com> wrote:
That's fair, if the primary focus of using the codec is purely to overcome a packet loss issue; however, consistent 30% packet loss to your endpoints seems like a fairly extreme scenario.
I suppose if you anticipate a high enough level of packet loss on the average call to justify the slightly degraded audio quality that will be introduced by transcoding, then it may be worthwhile.
I'll stand by the rest of my points though!
Best, Colin
On Mon, May 1, 2017 at 4:13 PM, Michael Jerris <mike@jerris.com> wrote:
I disagree… plenty of situations where you have a known quality leg to a provider over g711, and a uncontrolled internet leg to customer that you want to use opus/fec for.
On May 1, 2017, at 4:07 PM, Colin Morelli <colin.morelli@gmail.com> wrote:
Raul, I'm sure the OP was referring to 30% packet loss with FEC, which opus can turn back into a usable audio stream. You'll absolutely notice artifacts in the audio, and you'd clearly identify something is wrong, but it's a heck of a lot better than 30% packet loss without FEC.
Colton,
Freeswitch supports opus, yes. I'd be curious to know why you want to transcode between opus and G.711, unless your endpoints don't support G.711 though. Yes, FEC can help you with packet loss on one end, but if your packet loss is on the G.711 end, then you're SOL anyway, so it seems pointless to do this unless you can use opus all the way though.
As to how it will sound if doing the conversion, well, you're transcoding from one lossy codec to another lossy codec. The end result will be audio that will not sound as good as if it were just encoded in G.711 to begin with. Remember, the nature of lossy codecs is that they're, well, lossy. You can't make a G.711 stream sound better by re-encoding it to opus. Once it's encoded in G.711, that information is lost.
As for CPU usage, opus is not particularly cheap. Transcoding + resampling (if required), will considerably limit the number of concurrent calls your FS instance can handle. Actual results are going to depend on a number of factors that make it infeasible to talk about it. But again, you probably don't want to be transcoding between these two codecs unless you absolutely have two (i.e. you can't get the two endpoints to otherwise agree on a codec). You're not going to gain any real advantage from doing this.
Best, Colin
On Mon, May 1, 2017 at 3:54 PM, Raúl Alexis Betancor Santana < rbetancor@gmail.com> wrote:
30% of packet loss ... are you mad? ... there is no way any codec could recover from that. Most of codecs, you will notice whenever you have more than 1-3% packet loss ... 30% ? ... really? ... should sound like and old synthetizer
2017-05-01 20:03 GMT+01:00 Colton Conor <colton.conor@gmail.com>:
Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny?
We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711.
How much processing power would it take to transcode from OPUS to G711?
How well is the OPUS codec implemented on newer IP phones?
I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it?
Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram.
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Instead of trying to go HD with Opus, when you know there will be transcoding with G.711, you may consider using Opus at 8 KHz (assuming your devices can). This is supported by FS and in particular advisable if the clients are mobile devices. It will also reduce the bandwidth usage. With that packet loss, FEC becomes crucial. We documented various considerations in this area from a previous project: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus... For what concerns the estimated capacity, I have the same feeling as Colin above and you'll have to do your load testing. Best Regards, Giacomo On 1 May 2017 at 23:39, Colin Morelli <colin.morelli@gmail.com> wrote:
I agree, and was wrong before to suggest that there's no real advantage. FEC can be helpful to combat packet loss on unpredictable networks. Although again, I'll stand by my statement that consistent 10% packet loss would be pretty significant on any modern internet connection. Even consistent 2% packet loss would be cause for some concern. Intermittent packet loss issues, sure; but sustained 10% packet loss would be fairly alarming to hear about on any large scale. At that level, it's likely the user has problems with a number of services (even TCP ones, as they're likely to be experiencing slower speeds). On a consumer internet connection, your more frequent issues are almost certainly going to be latency and jitter, and generally your first line of defense (at least for jitter) would be the endpoints' jitter buffers. Although FEC can also help if you have to start dropping packets from the jitter buffer.
Again, I'm not trying to claim it can't or won't happen. It can, and opus will help when it does. I'm just saying it probably makes sense to actually confirm this is the issue (if you're currently experiencing problems). Maybe my experiences have been different.
*Are you saying transcoding from OPUS client to G711 is going to sound worse than just talking G711 all the way through? *
Yes. Both codecs are lossy. Any time you encode to a lossy codec, you lose data. In this case, you're doing it twice. The originating device will be encoding raw audio to G.711, and FS will decode the G.711 and re-encode as opus. The degradation on the transcoding won't be huge, and it many cases it may not even really be noticeable (you'd have to test this in your own setup to see), but it will exist.
I won't try to answer the question of how many transcoding channels any particular server can handle though. Perhaps the FS guys will have a decent answer, but I think they'll probably give similar feedback - you just need to test and find out. There are so many factors that come into play with performance that it becomes painfully difficult to try to guess.
Best, Colin
On Mon, May 1, 2017 at 5:21 PM, Colton Conor <colton.conor@gmail.com> wrote:
I am just judging the 30 percent packet loss with FEC. Example are on the OPUS website https://opus-codec.org/examples/ Sounds pretty good to me. I would be terrified to know what G711 sounds like without FEC having evien 10 percent packet loss.
I think there are many cases where one leg of the audio is uncontrolled, and the other leg of the audio via G711 is stable. For example, a freeswitch server sitting in a data center with redundant gigabit internet connections connected to a origination/termination carrier via G711.
Then clients using OTT internet connections like their home cable connection, LTE, or wifi. The clients are the leg I want in OPUS to communicate back to the freeswitch server. Yes, the clients are able to speak G711 too, but I want OPUS for the FEC and bandwidth control.
Are you saying transcoding from OPUS client to G711 is going to sound worse than just talking G711 all the way through?
So what can a new Dell server with most recent gen intel xeon processors handle as far as transcoding goes?
On Mon, May 1, 2017 at 3:19 PM, Colin Morelli <colin.morelli@gmail.com> wrote:
That's fair, if the primary focus of using the codec is purely to overcome a packet loss issue; however, consistent 30% packet loss to your endpoints seems like a fairly extreme scenario.
I suppose if you anticipate a high enough level of packet loss on the average call to justify the slightly degraded audio quality that will be introduced by transcoding, then it may be worthwhile.
I'll stand by the rest of my points though!
Best, Colin
On Mon, May 1, 2017 at 4:13 PM, Michael Jerris <mike@jerris.com> wrote:
I disagree… plenty of situations where you have a known quality leg to a provider over g711, and a uncontrolled internet leg to customer that you want to use opus/fec for.
On May 1, 2017, at 4:07 PM, Colin Morelli <colin.morelli@gmail.com> wrote:
Raul, I'm sure the OP was referring to 30% packet loss with FEC, which opus can turn back into a usable audio stream. You'll absolutely notice artifacts in the audio, and you'd clearly identify something is wrong, but it's a heck of a lot better than 30% packet loss without FEC.
Colton,
Freeswitch supports opus, yes. I'd be curious to know why you want to transcode between opus and G.711, unless your endpoints don't support G.711 though. Yes, FEC can help you with packet loss on one end, but if your packet loss is on the G.711 end, then you're SOL anyway, so it seems pointless to do this unless you can use opus all the way though.
As to how it will sound if doing the conversion, well, you're transcoding from one lossy codec to another lossy codec. The end result will be audio that will not sound as good as if it were just encoded in G.711 to begin with. Remember, the nature of lossy codecs is that they're, well, lossy. You can't make a G.711 stream sound better by re-encoding it to opus. Once it's encoded in G.711, that information is lost.
As for CPU usage, opus is not particularly cheap. Transcoding + resampling (if required), will considerably limit the number of concurrent calls your FS instance can handle. Actual results are going to depend on a number of factors that make it infeasible to talk about it. But again, you probably don't want to be transcoding between these two codecs unless you absolutely have two (i.e. you can't get the two endpoints to otherwise agree on a codec). You're not going to gain any real advantage from doing this.
Best, Colin
On Mon, May 1, 2017 at 3:54 PM, Raúl Alexis Betancor Santana < rbetancor@gmail.com> wrote:
30% of packet loss ... are you mad? ... there is no way any codec could recover from that. Most of codecs, you will notice whenever you have more than 1-3% packet loss ... 30% ? ... really? ... should sound like and old synthetizer
2017-05-01 20:03 GMT+01:00 Colton Conor <colton.conor@gmail.com>:
Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny?
We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711.
How much processing power would it take to transcode from OPUS to G711?
How well is the OPUS codec implemented on newer IP phones?
I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it?
Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram.
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Yealink is supporting OPUS on several other models since firmware V81: Quote: "We will support opus on the standard V81 of SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." Source: http://forum.yealink.com/forum/showthread.php?tid=29650&pid=51262&mode=threa... I can confirm OPUS implementation on the 'T21P E2' model. -- EL
I've reached 2000 simultaneous calls on a Dell Blade showing 40 CPU cores in /proc/cpuinfo , transcoding opus@8000h <-> PCMA with FS. Two years ago. If you want to encode FEC you'll need to use opus@8000h as codec_string or tweak the config settings in the xml to set the encoder in Narrowband audio . opus@16000h (newly added) should work with encoding FEC too. When you prepare your setup use CLI command opus_debug and netem to introduce packet loss, this way you'll be sure if the phones are sending FEC or not. Some may try to vary the bitrate to a value where the opus encoder simply does not encode FEC. Freeswitch, just like WebRTC stack sets by default the encoder in Fullband , which is CELT mode, and there's no FEC there. It's all in the doc Giacomo shown. Cheers, Dragos On Wed, May 3, 2017 at 8:16 AM, EL <fs.user@fordior.net> wrote:
Yealink is supporting OPUS on several other models since firmware V81:
Quote: "We will support opus on the standard V81 of SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2."
Source: http://forum.yealink.com/forum/showthread.php?tid= 29650&pid=51262&mode=threaded
I can confirm OPUS implementation on the 'T21P E2' model.
-- EL
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None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS support, but again no FEC/PLC support. /b On Wed, May 3, 2017 at 2:16 AM, EL <fs.user@fordior.net> wrote:
Yealink is supporting OPUS on several other models since firmware V81:
Quote: "We will support opus on the standard V81 of SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2."
Source: http://forum.yealink.com/forum/showthread.php?tid= 29650&pid=51262&mode=threaded
I can confirm OPUS implementation on the 'T21P E2' model.
-- EL
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So what does today support FEC/PLC today? We tested two Polycom VVX phones with the opus codec, and overall thought that G.722 sounded much better than OPUS for some reason. So using Opus at 8 KHz, does that mean that there will be less or no transcoding to G.711 since its also in 8 KHz? What are most of the web only WebRTC companies using as far as OPUS goes? What bitrate and KHz? On Fri, May 5, 2017 at 8:33 AM, Brian West <brian@freeswitch.org> wrote:
None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS support, but again no FEC/PLC support.
/b
On Wed, May 3, 2017 at 2:16 AM, EL <fs.user@fordior.net> wrote:
Yealink is supporting OPUS on several other models since firmware V81:
Quote: "We will support opus on the standard V81 of SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2."
Source: http://forum.yealink.com/forum/showthread.php?tid=29650&pid= 51262&mode=threaded
I can confirm OPUS implementation on the 'T21P E2' model.
-- EL
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Transcoding != resampling. Often times you will have to do both to convert audio streams, where transcoding is the act of converting between the two audio codecs' data format, and resampling is changing the sampling rate of the audio (8khz in your case). Opus is a completely different audio codec than G.711. Even if they're both at 8 khz, you'll have to transcode between the two, though you may be able to avoid the cost of resampling the audio to a different rate. Opus can handle sampling rates from 8khz-48khz. Sampling rates make a *huge* difference to audio quality (they'll effectively define the range of audio frequencies you can capture). If you're running opus at a constant 8khz, the reason G.722 sounds better is almost certainly because it's using a 16khz sampling rate. Most likely if you run opus at 16khz, it'll sound similar to G.722. If you have the capacity (and the devices support it), pushing opus to super-wideband or full band (24khz/48khz sampling rates, respectively) will yield much better audio quality. My guess (admittedly it's just an educated assumption), is that most WebRTC companies are using fullband opus in VBR mode simply because this is the default in WebRTC. Without mangling the SDP, this is what you're going to get. Best, Colin On Sat, May 6, 2017 at 10:41 AM, Colton Conor <colton.conor@gmail.com> wrote:
So what does today support FEC/PLC today?
We tested two Polycom VVX phones with the opus codec, and overall thought that G.722 sounded much better than OPUS for some reason.
So using Opus at 8 KHz, does that mean that there will be less or no transcoding to G.711 since its also in 8 KHz?
What are most of the web only WebRTC companies using as far as OPUS goes? What bitrate and KHz?
On Fri, May 5, 2017 at 8:33 AM, Brian West <brian@freeswitch.org> wrote:
None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS support, but again no FEC/PLC support.
/b
On Wed, May 3, 2017 at 2:16 AM, EL <fs.user@fordior.net> wrote:
Yealink is supporting OPUS on several other models since firmware V81:
Quote: "We will support opus on the standard V81 of SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2."
Source: http://forum.yealink.com/forum/showthread.php?tid=29650&pid= 51262&mode=threaded
I can confirm OPUS implementation on the 'T21P E2' model.
-- EL
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FreeSWITCH can limit Opus to a configured sample rate but the OPUS specification says that the SDP must always specify 48 kHz and 2 channels. OPUS is a combination of SILK and CELT, with the original PLC+FEC coming from SILK. There are effectively three modes in OPUS - SILK, CELT, and hybrid. It's not clear to me how well PLC+FEC works in the hybrid and CELT modes used at higher sample rates. I haven't looked at it specifically (or tested it) but my suspicion is that PLC+FEC works much better the closer you get to "pure" SILK mode, which is essentially what limiting the sample rate to 8 kHz does. On Sat, May 6, 2017 at 10:21 AM, Colin Morelli <colin.morelli@gmail.com> wrote:
Transcoding != resampling. Often times you will have to do both to convert audio streams, where transcoding is the act of converting between the two audio codecs' data format, and resampling is changing the sampling rate of the audio (8khz in your case). Opus is a completely different audio codec than G.711. Even if they're both at 8 khz, you'll have to transcode between the two, though you may be able to avoid the cost of resampling the audio to a different rate.
Opus can handle sampling rates from 8khz-48khz. Sampling rates make a huge difference to audio quality (they'll effectively define the range of audio frequencies you can capture). If you're running opus at a constant 8khz, the reason G.722 sounds better is almost certainly because it's using a 16khz sampling rate. Most likely if you run opus at 16khz, it'll sound similar to G.722. If you have the capacity (and the devices support it), pushing opus to super-wideband or full band (24khz/48khz sampling rates, respectively) will yield much better audio quality.
My guess (admittedly it's just an educated assumption), is that most WebRTC companies are using fullband opus in VBR mode simply because this is the default in WebRTC. Without mangling the SDP, this is what you're going to get.
Best, Colin
On Sat, May 6, 2017 at 10:41 AM, Colton Conor <colton.conor@gmail.com> wrote:
So what does today support FEC/PLC today?
We tested two Polycom VVX phones with the opus codec, and overall thought that G.722 sounded much better than OPUS for some reason.
So using Opus at 8 KHz, does that mean that there will be less or no transcoding to G.711 since its also in 8 KHz?
What are most of the web only WebRTC companies using as far as OPUS goes? What bitrate and KHz?
On Fri, May 5, 2017 at 8:33 AM, Brian West <brian@freeswitch.org> wrote:
None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS support, but again no FEC/PLC support.
/b
On Wed, May 3, 2017 at 2:16 AM, EL <fs.user@fordior.net> wrote:
Yealink is supporting OPUS on several other models since firmware V81:
Quote: "We will support opus on the standard V81 of SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2."
Source: http://forum.yealink.com/forum/showthread.php?tid=29650&pid=51262&mode=threa...
I can confirm OPUS implementation on the 'T21P E2' model.
-- EL
_________________________________________________________________________ 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
Book a phone call (CST)
Allison prompts for FreeSWITCH:
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Got Bugs? Report them here! | Reddit: /r/freeswitch
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest
_________________________________________________________________________ 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
-- Kristian Kielhofner
Anyone know on Polycom VVX's using OPUS, what is the sample rate? 8Khz or 48Khz or somewhere inbetween? I looked a at call through VoIP Monitor, and it said OPUS48 as the codec. Buy since "the OPUS specification says that the SDP must always specify 48 kHz and 2 channels" according to Kristian Kielhofner", I assume it just sadys OPUS 48 because of this. Like I mentioned, I though that G.722 between two Polycom's sounded better than OPUS to my suprise. I assume this is because G.722 is at 16 kHZ, and Polycom's opus is potentially at 8kHZ. The softswitch we are using (Netsapiens) does not do any trasconding to my knowledge, so the payload has to be whatever Polycom encodes/decodes at. Is there a way to tell what kHz the Opus codec is running at on a Polcom VVX, or by looked at a packet capture? On Sat, May 6, 2017 at 1:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
FreeSWITCH can limit Opus to a configured sample rate but the OPUS specification says that the SDP must always specify 48 kHz and 2 channels.
OPUS is a combination of SILK and CELT, with the original PLC+FEC coming from SILK. There are effectively three modes in OPUS - SILK, CELT, and hybrid. It's not clear to me how well PLC+FEC works in the hybrid and CELT modes used at higher sample rates. I haven't looked at it specifically (or tested it) but my suspicion is that PLC+FEC works much better the closer you get to "pure" SILK mode, which is essentially what limiting the sample rate to 8 kHz does.
On Sat, May 6, 2017 at 10:21 AM, Colin Morelli <colin.morelli@gmail.com> wrote:
Transcoding != resampling. Often times you will have to do both to convert audio streams, where transcoding is the act of converting between the two audio codecs' data format, and resampling is changing the sampling rate of the audio (8khz in your case). Opus is a completely different audio codec than G.711. Even if they're both at 8 khz, you'll have to transcode between the two, though you may be able to avoid the cost of resampling the audio to a different rate.
Opus can handle sampling rates from 8khz-48khz. Sampling rates make a huge difference to audio quality (they'll effectively define the range of audio frequencies you can capture). If you're running opus at a constant 8khz, the reason G.722 sounds better is almost certainly because it's using a 16khz sampling rate. Most likely if you run opus at 16khz, it'll sound similar to G.722. If you have the capacity (and the devices support it), pushing opus to super-wideband or full band (24khz/48khz sampling rates, respectively) will yield much better audio quality.
My guess (admittedly it's just an educated assumption), is that most WebRTC companies are using fullband opus in VBR mode simply because this is the default in WebRTC. Without mangling the SDP, this is what you're going to get.
Best, Colin
On Sat, May 6, 2017 at 10:41 AM, Colton Conor <colton.conor@gmail.com> wrote:
So what does today support FEC/PLC today?
We tested two Polycom VVX phones with the opus codec, and overall
thought
that G.722 sounded much better than OPUS for some reason.
So using Opus at 8 KHz, does that mean that there will be less or no transcoding to G.711 since its also in 8 KHz?
What are most of the web only WebRTC companies using as far as OPUS goes? What bitrate and KHz?
On Fri, May 5, 2017 at 8:33 AM, Brian West <brian@freeswitch.org> wrote:
None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS support, but again no FEC/PLC support.
/b
On Wed, May 3, 2017 at 2:16 AM, EL <fs.user@fordior.net> wrote:
Yealink is supporting OPUS on several other models since firmware V81:
Quote: "We will support opus on the standard V81 of SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2."
29650&pid=51262&mode=threaded
I can confirm OPUS implementation on the 'T21P E2' model.
-- EL
____________________________________________________________
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
--
Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Book a phone call (CST)
Allison prompts for FreeSWITCH:
https://www.gofundme.com/allison-prompts-for-freeswitch
Got Bugs? Report them here! | Reddit: /r/freeswitch
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest
____________________________________________________________
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
-- Kristian Kielhofner
_________________________________________________________________________ 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
Better check with "opus_debug on" on the FS CLI. Eg: 2017-05-10 16:28:05.724001 [DEBUG] mod_opus.c:453 decode: opus_frames [1] samples [960] audio bandwidth [WIDEBAND] bytes [75] FEC[yes] channels[1] Look at the audio bandwidth of the decoded packets . If the audio bandwitdh is WIDEBAND, then the sample rate is 16 khz. You cannot find this out by looking at the pcap , the RTP timestamp is incremented as described by the SDP profile , so always at 48 khz (which is 960 for 20 ms ptime ) . Cheers, Dragos On Wed, May 10, 2017 at 2:38 PM, Colton Conor <colton.conor@gmail.com> wrote:
Anyone know on Polycom VVX's using OPUS, what is the sample rate? 8Khz or 48Khz or somewhere inbetween?
I looked a at call through VoIP Monitor, and it said OPUS48 as the codec. Buy since "the OPUS specification says that the SDP must always specify 48 kHz and 2 channels" according to Kristian Kielhofner", I assume it just sadys OPUS 48 because of this.
Like I mentioned, I though that G.722 between two Polycom's sounded better than OPUS to my suprise. I assume this is because G.722 is at 16 kHZ, and Polycom's opus is potentially at 8kHZ. The softswitch we are using (Netsapiens) does not do any trasconding to my knowledge, so the payload has to be whatever Polycom encodes/decodes at.
Is there a way to tell what kHz the Opus codec is running at on a Polcom VVX, or by looked at a packet capture?
On Sat, May 6, 2017 at 1:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
FreeSWITCH can limit Opus to a configured sample rate but the OPUS specification says that the SDP must always specify 48 kHz and 2 channels.
OPUS is a combination of SILK and CELT, with the original PLC+FEC coming from SILK. There are effectively three modes in OPUS - SILK, CELT, and hybrid. It's not clear to me how well PLC+FEC works in the hybrid and CELT modes used at higher sample rates. I haven't looked at it specifically (or tested it) but my suspicion is that PLC+FEC works much better the closer you get to "pure" SILK mode, which is essentially what limiting the sample rate to 8 kHz does.
On Sat, May 6, 2017 at 10:21 AM, Colin Morelli <colin.morelli@gmail.com> wrote:
Transcoding != resampling. Often times you will have to do both to convert audio streams, where transcoding is the act of converting between the two audio codecs' data format, and resampling is changing the sampling rate of the audio (8khz in your case). Opus is a completely different audio codec than G.711. Even if they're both at 8 khz, you'll have to transcode between the two, though you may be able to avoid the cost of resampling the audio to a different rate.
Opus can handle sampling rates from 8khz-48khz. Sampling rates make a huge difference to audio quality (they'll effectively define the range of audio frequencies you can capture). If you're running opus at a constant 8khz, the reason G.722 sounds better is almost certainly because it's using a 16khz sampling rate. Most likely if you run opus at 16khz, it'll sound similar to G.722. If you have the capacity (and the devices support it), pushing opus to super-wideband or full band (24khz/48khz sampling rates, respectively) will yield much better audio quality.
My guess (admittedly it's just an educated assumption), is that most WebRTC companies are using fullband opus in VBR mode simply because this is the default in WebRTC. Without mangling the SDP, this is what you're going to get.
Best, Colin
On Sat, May 6, 2017 at 10:41 AM, Colton Conor <colton.conor@gmail.com> wrote:
So what does today support FEC/PLC today?
We tested two Polycom VVX phones with the opus codec, and overall
thought
that G.722 sounded much better than OPUS for some reason.
So using Opus at 8 KHz, does that mean that there will be less or no transcoding to G.711 since its also in 8 KHz?
What are most of the web only WebRTC companies using as far as OPUS goes? What bitrate and KHz?
On Fri, May 5, 2017 at 8:33 AM, Brian West <brian@freeswitch.org> wrote:
None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS support, but again no FEC/PLC support.
/b
On Wed, May 3, 2017 at 2:16 AM, EL <fs.user@fordior.net> wrote:
Yealink is supporting OPUS on several other models since firmware V81:
Quote: "We will support opus on the standard V81 of SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2."
Source: http://forum.yealink.com/forum/showthread.php?tid=29650&pid=
51262&mode=threaded
I can confirm OPUS implementation on the 'T21P E2' model.
-- EL
____________________________________________________________
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
--
Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Book a phone call (CST)
Allison prompts for FreeSWITCH:
https://www.gofundme.com/allison-prompts-for-freeswitch
Got Bugs? Report them here! | Reddit: /r/freeswitch
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest
____________________________________________________________
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
-- Kristian Kielhofner
_________________________________________________________________________ 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
Just because "according to Kristian Kielhofner" sounds strange to me; how about according to RFC 7587 :)? https://tools.ietf.org/html/rfc7587#section-7 "The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding name. The RTP clock rate in "a=rtpmap" MUST be 48000, and the number of channels MUST be 2." On Wed, May 10, 2017 at 8:38 AM, Colton Conor <colton.conor@gmail.com> wrote:
Anyone know on Polycom VVX's using OPUS, what is the sample rate? 8Khz or 48Khz or somewhere inbetween?
I looked a at call through VoIP Monitor, and it said OPUS48 as the codec. Buy since "the OPUS specification says that the SDP must always specify 48 kHz and 2 channels" according to Kristian Kielhofner", I assume it just sadys OPUS 48 because of this.
Like I mentioned, I though that G.722 between two Polycom's sounded better than OPUS to my suprise. I assume this is because G.722 is at 16 kHZ, and Polycom's opus is potentially at 8kHZ. The softswitch we are using (Netsapiens) does not do any trasconding to my knowledge, so the payload has to be whatever Polycom encodes/decodes at.
Is there a way to tell what kHz the Opus codec is running at on a Polcom VVX, or by looked at a packet capture?
On Sat, May 6, 2017 at 1:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
FreeSWITCH can limit Opus to a configured sample rate but the OPUS specification says that the SDP must always specify 48 kHz and 2 channels.
OPUS is a combination of SILK and CELT, with the original PLC+FEC coming from SILK. There are effectively three modes in OPUS - SILK, CELT, and hybrid. It's not clear to me how well PLC+FEC works in the hybrid and CELT modes used at higher sample rates. I haven't looked at it specifically (or tested it) but my suspicion is that PLC+FEC works much better the closer you get to "pure" SILK mode, which is essentially what limiting the sample rate to 8 kHz does.
On Sat, May 6, 2017 at 10:21 AM, Colin Morelli <colin.morelli@gmail.com> wrote:
Transcoding != resampling. Often times you will have to do both to convert audio streams, where transcoding is the act of converting between the two audio codecs' data format, and resampling is changing the sampling rate of the audio (8khz in your case). Opus is a completely different audio codec than G.711. Even if they're both at 8 khz, you'll have to transcode between the two, though you may be able to avoid the cost of resampling the audio to a different rate.
Opus can handle sampling rates from 8khz-48khz. Sampling rates make a huge difference to audio quality (they'll effectively define the range of audio frequencies you can capture). If you're running opus at a constant 8khz, the reason G.722 sounds better is almost certainly because it's using a 16khz sampling rate. Most likely if you run opus at 16khz, it'll sound similar to G.722. If you have the capacity (and the devices support it), pushing opus to super-wideband or full band (24khz/48khz sampling rates, respectively) will yield much better audio quality.
My guess (admittedly it's just an educated assumption), is that most WebRTC companies are using fullband opus in VBR mode simply because this is the default in WebRTC. Without mangling the SDP, this is what you're going to get.
Best, Colin
On Sat, May 6, 2017 at 10:41 AM, Colton Conor <colton.conor@gmail.com> wrote:
So what does today support FEC/PLC today?
We tested two Polycom VVX phones with the opus codec, and overall thought that G.722 sounded much better than OPUS for some reason.
So using Opus at 8 KHz, does that mean that there will be less or no transcoding to G.711 since its also in 8 KHz?
What are most of the web only WebRTC companies using as far as OPUS goes? What bitrate and KHz?
On Fri, May 5, 2017 at 8:33 AM, Brian West <brian@freeswitch.org> wrote:
None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS support, but again no FEC/PLC support.
/b
On Wed, May 3, 2017 at 2:16 AM, EL <fs.user@fordior.net> wrote:
Yealink is supporting OPUS on several other models since firmware V81:
Quote: "We will support opus on the standard V81 of SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2."
Source:
http://forum.yealink.com/forum/showthread.php?tid=29650&pid=51262&mode=threa...
I can confirm OPUS implementation on the 'T21P E2' model.
-- EL
_________________________________________________________________________ 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
--
Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Book a phone call (CST)
Allison prompts for FreeSWITCH:
https://www.gofundme.com/allison-prompts-for-freeswitch
Got Bugs? Report them here! | Reddit: /r/freeswitch
T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest
_________________________________________________________________________ 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
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-- Kristian Kielhofner
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_________________________________________________________________________ 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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-- Kristian Kielhofner
On May 10, 2017, at 12:40 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Just because "according to Kristian Kielhofner" sounds strange to me; how about according to RFC 7587 :)?
https://tools.ietf.org/html/rfc7587#section-7
"The media subtype ("opus") goes in SDP "a=rtpmap" as the encoding name. The RTP clock rate in "a=rtpmap" MUST be 48000, and the number of channels MUST be 2.”
- According to Kristian Kielhofner and RFC 7587….
On Wed, May 10, 2017 at 8:38 AM, Colton Conor <colton.conor@gmail.com> wrote:
Anyone know on Polycom VVX's using OPUS, what is the sample rate? 8Khz or 48Khz or somewhere inbetween?
I looked a at call through VoIP Monitor, and it said OPUS48 as the codec. Buy since "the OPUS specification says that the SDP must always specify 48 kHz and 2 channels" according to Kristian Kielhofner", I assume it just sadys OPUS 48 because of this.
Like I mentioned, I though that G.722 between two Polycom's sounded better than OPUS to my suprise. I assume this is because G.722 is at 16 kHZ, and Polycom's opus is potentially at 8kHZ. The softswitch we are using (Netsapiens) does not do any trasconding to my knowledge, so the payload has to be whatever Polycom encodes/decodes at.
Is there a way to tell what kHz the Opus codec is running at on a Polcom VVX, or by looked at a packet capture?
On Sat, May 6, 2017 at 1:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
FreeSWITCH can limit Opus to a configured sample rate but the OPUS specification says that the SDP must always specify 48 kHz and 2 channels.
OPUS is a combination of SILK and CELT, with the original PLC+FEC coming from SILK. There are effectively three modes in OPUS - SILK, CELT, and hybrid. It's not clear to me how well PLC+FEC works in the hybrid and CELT modes used at higher sample rates. I haven't looked at it specifically (or tested it) but my suspicion is that PLC+FEC works much better the closer you get to "pure" SILK mode, which is essentially what limiting the sample rate to 8 kHz does.
On Sat, May 6, 2017 at 10:21 AM, Colin Morelli <colin.morelli@gmail.com> wrote:
Transcoding != resampling. Often times you will have to do both to convert audio streams, where transcoding is the act of converting between the two audio codecs' data format, and resampling is changing the sampling rate of the audio (8khz in your case). Opus is a completely different audio codec than G.711. Even if they're both at 8 khz, you'll have to transcode between the two, though you may be able to avoid the cost of resampling the audio to a different rate.
Opus can handle sampling rates from 8khz-48khz. Sampling rates make a huge difference to audio quality (they'll effectively define the range of audio frequencies you can capture). If you're running opus at a constant 8khz, the reason G.722 sounds better is almost certainly because it's using a 16khz sampling rate. Most likely if you run opus at 16khz, it'll sound similar to G.722. If you have the capacity (and the devices support it), pushing opus to super-wideband or full band (24khz/48khz sampling rates, respectively) will yield much better audio quality.
My guess (admittedly it's just an educated assumption), is that most WebRTC companies are using fullband opus in VBR mode simply because this is the default in WebRTC. Without mangling the SDP, this is what you're going to get.
Best, Colin
On Sat, May 6, 2017 at 10:41 AM, Colton Conor <colton.conor@gmail.com> wrote:
So what does today support FEC/PLC today?
We tested two Polycom VVX phones with the opus codec, and overall thought that G.722 sounded much better than OPUS for some reason.
So using Opus at 8 KHz, does that mean that there will be less or no transcoding to G.711 since its also in 8 KHz?
What are most of the web only WebRTC companies using as far as OPUS goes? What bitrate and KHz?
On Fri, May 5, 2017 at 8:33 AM, Brian West <brian@freeswitch.org> wrote:
None of the hardware phones do FEC/PLC last I tested, Sonus has OPUS support, but again no FEC/PLC support.
/b
On Wed, May 3, 2017 at 2:16 AM, EL <fs.user@fordior.net> wrote: > > > Yealink is supporting OPUS on several other models since firmware > V81: > > Quote: > "We will support opus on the standard V81 of > SIP-T40P/T23P/T23G/T2 1(P) E2/T19(P) E2." > > Source: > > http://forum.yealink.com/forum/showthread.php?tid=29650&pid=51262&mode=threa... > > I can confirm OPUS implementation on the 'T21P E2' model.
participants (9)
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Brian West
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Colin Morelli
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Colton Conor
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EL
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Giacomo Vacca
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Kristian Kielhofner
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Michael Jerris
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Oancea, Dragos
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Raúl Alexis Betancor Santana