Re: [asterisk-users] Different codecs for reading and writing
Philipp Kempgen schrieb: > Elliot Murdock schrieb: > >> I am wondering how the Asterisk community has been working on >> solutions to deal with the asymmetric quality of ADSL. Voip is >> becoming popular and a bottleneck does exists on the ADSL upload side. >> > > One participant's upload is the other participant's download and > vice-versa. So how would different codecs for sending and receiving > help? > Given that the other party does not use an asymetric internet connection it could actually help. Not that I would recomment such a mode over just using a low bandwidth codec like the aforementioned. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
On Mon, 3 Aug 2009, Elliot Murdock wrote: > Hello Everyone! > > Thank you for all the information. > > I am wondering how the Asterisk community has been working on > solutions to deal with the asymmetric quality of ADSL. Voip is > becoming popular and a bottleneck does exists on the ADSL upload side. What bottleneck? Distance limitations aside for *DSL technologies, I think what we get where I am is more than enough for your average SME. 830Kb/sec upload speed is OK for 10 concurrent G711 calls or a few dozen G729 calls. My view is that if a business really does need more concurrent calls than that, then there are alternative technologies avalable which are more suited to the larger enterprise - SDSL where avalable, or traditional 2Mb leased lines, or 10Mb LAN extension type circuits. And lets not forget that the ADSL backhaul is a contended service (at least where I am). You're not going to get any guarantees all that 830Kb/sec is going to be avalable all the time, but where you are things may well be different. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
Elliot Murdock schrieb: > I am wondering how the Asterisk community has been working on > solutions to deal with the asymmetric quality of ADSL. Voip is > becoming popular and a bottleneck does exists on the ADSL upload side. One participant's upload is the other participant's download and vice-versa. So how would different codecs for sending and receiving help? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
I'm not sure there IS an issue, per se. There are lower bitrate codecs that will work fine for voice communications in both directions. But if you're trying to force a low-end codec to the upstream, that just means the downstream on the remote end is going to be stuck with a low-end codec. And if he's trying to force a low-end codec on his upstream, then you're still going to get a low-end codec on your downstream. If you're relying on the Asterisk box somewhere in the middle to transcode these two streams into a higher bitrate codec, you're still not going to get any higher quality than you send. Your option is using a low bitrate codec for both directions and not bothering to try and min/max your streams. Excellent quality low bitrate codecs exist in the form of g729, iLBC, and Speex (in that order of prevalence). Just deal with the fact that, even if you min/maxed the streams, you still wouldn't be able to get any more streams than will fit in your upstream pipe, so let that be your guide for the technology involved. You may end up with some extra bandwidth on the downstream side, but trying to fill it up with something just to fill it up with something won't get you anywhere. N. Elliot Murdock wrote: > Hello Everyone! > > Thank you for all the information. > > I am wondering how the Asterisk community has been working on > solutions to deal with the asymmetric quality of ADSL. Voip is > becoming popular and a bottleneck does exists on the ADSL upload side. > > Elliot > > On Sun, Aug 2, 2009 at 3:17 PM, Kevin P. Fleming wrote: > >> Tim Panton wrote: >> >> >>> The protocol expects the 2 ends to agree a single symmetrical codec >>> as part of the connection setup, but it doesn't define what actually >>> happens >>> if the codec specified in the first (full frame) voice packet isn't what >>> was agreed. >>> >> Asterisk only supports symmetric codec configuration on its internal >> channels, so in Asterisk's IAX2 implementation, if a frame is received >> from the other endpoint that is not in the 'expected' format a warning >> is issued and the outbound direction is automatically switched to the >> same format. The same is done for any protocol using RTP in Asterisk. >> >> -- >> Kevin P. Fleming >> Digium, Inc. | Director of Software Technologies >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA >> skype: kpfleming | jabber: kpflem...@digium.com >> Check us out at www.digium.com & www.asterisk.org >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
Hello Everyone! Thank you for all the information. I am wondering how the Asterisk community has been working on solutions to deal with the asymmetric quality of ADSL. Voip is becoming popular and a bottleneck does exists on the ADSL upload side. Elliot On Sun, Aug 2, 2009 at 3:17 PM, Kevin P. Fleming wrote: > Tim Panton wrote: > >> The protocol expects the 2 ends to agree a single symmetrical codec >> as part of the connection setup, but it doesn't define what actually >> happens >> if the codec specified in the first (full frame) voice packet isn't what >> was agreed. > > Asterisk only supports symmetric codec configuration on its internal > channels, so in Asterisk's IAX2 implementation, if a frame is received > from the other endpoint that is not in the 'expected' format a warning > is issued and the outbound direction is automatically switched to the > same format. The same is done for any protocol using RTP in Asterisk. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
Tim Panton wrote: > The protocol expects the 2 ends to agree a single symmetrical codec > as part of the connection setup, but it doesn't define what actually > happens > if the codec specified in the first (full frame) voice packet isn't what > was agreed. Asterisk only supports symmetric codec configuration on its internal channels, so in Asterisk's IAX2 implementation, if a frame is received from the other endpoint that is not in the 'expected' format a warning is issued and the outbound direction is automatically switched to the same format. The same is done for any protocol using RTP in Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
On 1 Aug 2009, at 22:26, Alex Balashov wrote: Elliot Murdock wrote: Thank you...do you know if IAX can do this? The reason for doing is this is to get over the adsl upload/download discrepancy. While G711 gives terrific quality, it is not always that feasible for the upload direction, which has much more limited bandwidth. Accordingly, it would be possible to use G729 for upload, but keep the higher quality codec, G711, for download. I do not know a great deal about IAX so I will defer to the experts for the definitive word on whether it is possible from the point of view of its formal protocol mechanics. However, poking around the various configuration options for IAX peers on voip-info.org and a few other places suggests that there is no option to do that with IAX, either. It's not really something 99.9% of VoIP users want to do. :-) I think you will find that it may work with Asterisk's IAX implementation. The protocol expects the 2 ends to agree a single symmetrical codec as part of the connection setup, but it doesn't define what actually happens if the codec specified in the first (full frame) voice packet isn't what was agreed. I have a vague memory that if the codec is one that is allowed, asterisk does 'the right thing' issues a warning and uses what it was given. But, as Alex says, there is no clear way to define this in the config files. You would probably do better to use Speex in both directions, but set the encoding quality (in codec.conf ) parameters to be different at the 2 ends. The speex decoder should at the far end should be fine with that. see http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
Elliot Murdock wrote: > Thank you...do you know if IAX can do this? > > The reason for doing is this is to get over the adsl upload/download > discrepancy. While G711 gives terrific quality, it is not always that > feasible for the upload direction, which has much more limited > bandwidth. Accordingly, it would be possible to use G729 for upload, > but keep the higher quality codec, G711, for download. I do not know a great deal about IAX so I will defer to the experts for the definitive word on whether it is possible from the point of view of its formal protocol mechanics. However, poking around the various configuration options for IAX peers on voip-info.org and a few other places suggests that there is no option to do that with IAX, either. It's not really something 99.9% of VoIP users want to do. :-) As far as the asymmetric upstream/downstream rates on ADSL, are you sure that you're solving the right problem? How many calls are you trying to run over such a circuit? If it's one or two or three, it's no big deal either way to do with G.711; it's 64 kbps (8 kB/s), or ~80ish kbps with Layer 2 framing overhead (for Ethernet; it may be a little more once PPP(oE) headers + ATM AAL5 encapsulation + other stuff frequently in use in DSL transport and aggregation architecture is factored in). Personally, I'd recommend that you just use some low-bitrate codec in both directions. If it's a good codec, you won't take a tragically significant hit on the quality on the receive side either, and if it's a bad codec, then the feasibility of conversation with the far end is going to be impacted because they can't hear you well. Opinions on which codec to use vary, but the only low-bit rate codec I've ever used that I've been even remotely satisfied with in terms of conversational quality is G.729. It really provides by far the most optimal intersection of bit rate and quality; it trims the call down to 8 kbps (8x less than clear-channel G.711!) and still sounds quite good. It uses some rather advanced CELP (code-excited linear prediction) techniques to refer to forms out of a table to achieve that effect. The downside - and it probably has to do with why G.729 is so good - is that all implementations are subject to royalties on software patents, so you can't have it for free with Asterisk; at least, you can't have it legally. It comes built into most commercial VoIP gear because the G.729 licensing cost is just baked into the retail price of the unit. But with open-source stuff, it costs money. However, your mileage may vary; you may want to try Speex or iLBC or whatever the cool kids are using. The downside there is that not a lot of VoIP trunking providers' gear supports these. For the most part, the commercial part of the ecosystem (i.e. beyond Asterisk) deals in G.711u/A and G.729A, although the G.722 wideband codec (for "high-definition" voice) is starting to get some rather serious attention. Cheers, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
Hello, Thank you...do you know if IAX can do this? The reason for doing is this is to get over the adsl upload/download discrepancy. While G711 gives terrific quality, it is not always that feasible for the upload direction, which has much more limited bandwidth. Accordingly, it would be possible to use G729 for upload, but keep the higher quality codec, G711, for download. Thanks, Elliot On Sat, Aug 1, 2009 at 11:06 PM, Alex Balashov wrote: > Elliot Murdock wrote: > >> I am wondering how to configure Asterisk and devices so I can use >> different codecs for upstream and downstream packets. > > You can't. With SIP as the channel technology, at least, the SDP > negotiation model demands a uniform codec from both sides. > > I guess there's nothing to stop two endpoints that both have multi-codec > capabilities from using different codecs for send and receive in > *principle*, but it would fall outside the standard implementational > behaviour of SIP endpoints. > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (678) 237-1775 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different codecs for reading and writing
Elliot Murdock wrote: > I am wondering how to configure Asterisk and devices so I can use > different codecs for upstream and downstream packets. You can't. With SIP as the channel technology, at least, the SDP negotiation model demands a uniform codec from both sides. I guess there's nothing to stop two endpoints that both have multi-codec capabilities from using different codecs for send and receive in *principle*, but it would fall outside the standard implementational behaviour of SIP endpoints. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Different codecs for reading and writing
Hello! I am wondering how to configure Asterisk and devices so I can use different codecs for upstream and downstream packets. Thank you, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
