I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP, SCCP and H323 come to mind)? Rather than asking for a full education here, can somebody point me at a suitable practical reference? Of course, if somebody wants to actually post the answer that'd be fine too :) THanks, </edg>
The quick tyrannical answer, Each T1 channels is 64Kbps, 1,544,000 / 24 = 64,333 G.711 CODEC is used on the T1 Channels. So if you use G.711 codec then you will be able run 24 SIP conversations on a T1. SIP is not a codec, SIP is a call control protocol. SIP does the work of connecting two endpoints together. MGCP, SCCP and H323 are also just call control protocols. SIP has low overhead while H323 has lots of features. The overhead is so small that it won't really figure in the CODEC calculations. The RTP protocol is responsible for moving the voice data from point to point. But I digress. When you look at the CODECs each compresses the voice data differently. It is this compression that gives you your number of "phone calls" on a T1. As per -- http://www.vocal.com/data_sheets/full/code_source_voip_g723.html Calls per T1 | Codec explanation 289 or 240 |?G.723 (often referred to as G.723.1) - 5 1/3k and 6.4k bps ACELP/MP-MLQ 193 |?G.729 - 8k bps CS-ACELP ?G.729A - reduced complexity version of G.729 - fewer MIPS at the expense of reduced perceived signal quality 118 |?GSM 06.10 - 13k bps RPE-LTP 96 |?G.728 - 16k bps LD-CELP 96 to 38 |?G.726 - 16k, 24k, 32k and 40k bps ADPCM - normally not used in Voice-over-IP applications 48 |?G.721 - 32k bps ADPCM - normally not used in Voice-over-IP applications 24 |?G.711 - 64k bps PCM (A-Law or ?-Law format) The above table shows one of the reasons G.729 is popular in that you can get 192 calls per T1 with fair quality. Remember time is money; the tighter the compression the more time it takes to compress/decompress and therefore the more money in silicon it takes to do the compressions on the fly. Smaller call "channel/bandwidth" means more hardware horsepower to compress and decompress the voice on the call. Race "The Tyrant" Van der Decken -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ed Greenberg Sent: 16 December 2004 12:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Calculating required bandwidth I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP, SCCP and H323 come to mind)? Rather than asking for a full education here, can somebody point me at a suitable practical reference? Of course, if somebody wants to actually post the answer that'd be fine too :) THanks, </edg> _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
A TDM channel (POTS/PRI) consists of 64k of real-time data, VoIP with no compression (g.711) consists of 64k plus IP protocol overhead for a total bandwidth or 80 to 90k required per uncompressed channel. So a IP T1 carrying VoIP without compression has lower capacity that a Voice T1. A t1 for voice typically carries 23 b channels and 1 d channel, so 23 conversations not 24. If you use compression on the VoIP traffic you gain capacity, but loose CPU performance as the RTP data stream has to be transcoded by *. If compression is used, and the box has the CPU power, significantly more than 23 is the answer, probably limited more by then number that your * can setup, transcode, and tear down. The exact answer depends on your use and can only be determined through testing. Uncompressed the answer is probably closer to 15 to 18 RTP streams across a dedicate T1 IP link.> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Ed Greenberg > Sent: Thursday, December 16, 2004 10:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Calculating required bandwidth > > I was posed this question: > > A T1 set up for voice carries 24 conversations on a circuit > that is 1.544 megabits/second. Right? > > Well, if you set that T1 up to carry data and run a link > between two IP networks over it, how many SIP conversations > could it be expected to carry? > How about IAX? > > How would one extend this calculation to varying bandwidth > circuits and various VOIP protocols (MGCP, SCCP and H323 come > to mind)? > > Rather than asking for a full education here, can somebody > point me at a suitable practical reference? Of course, if > somebody wants to actually post the answer that'd be fine too :) > > THanks, > </edg> > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption A T1 set up for voice uses G711 at 64Kbps which at 1.544Mbps equals 24 channels. This is also known as a PRI. -Matthew ----- Original Message ----- From: "Ed Greenberg" <edg@greenberg.org> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Thursday, December 16, 2004 11:45 AM Subject: [Asterisk-Users] Calculating required bandwidth> I was posed this question: > > A T1 set up for voice carries 24 conversations on a circuit that is 1.544 > megabits/second. Right? > > Well, if you set that T1 up to carry data and run a link between two IP > networks over it, how many SIP conversations could it be expected tocarry?> How about IAX? > > How would one extend this calculation to varying bandwidth circuits and > various VOIP protocols (MGCP, SCCP and H323 come to mind)? > > Rather than asking for a full education here, can somebody point me at a > suitable practical reference? Of course, if somebody wants to actuallypost> the answer that'd be fine too :) > > THanks, > </edg> > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
"So if you use G.711 codec then you will be able run 24 SIP conversations on a T1." Not true, the RTP stream is 64k, there is also IP packet and VoIP protocol overhead to deal with. If you try to dedicate less than 80k+ per g.711 stream you will have trouble and you also have IP data on the T1 you are really looking for trouble. Priority queuing will help, but only if you have enough bandwidth to transfer all of your RTP and data packets.> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Race Vanderdecken > Sent: Thursday, December 16, 2004 11:53 AM > To: 'Ed Greenberg'; 'Asterisk Users Mailing List - > Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Calculating required bandwidth > > The quick tyrannical answer, > > Each T1 channels is 64Kbps, 1,544,000 / 24 = 64,333 > > G.711 CODEC is used on the T1 Channels. > > So if you use G.711 codec then you will be able run 24 SIP > conversations on a T1. > > SIP is not a codec, SIP is a call control protocol. SIP does > the work of connecting two endpoints together. MGCP, SCCP and > H323 are also just call control protocols. SIP has low > overhead while H323 has lots of features. The overhead is so > small that it won't really figure in the CODEC calculations. > > The RTP protocol is responsible for moving the voice data > from point to point. But I digress. > > When you look at the CODECs each compresses the voice data > differently. It is this compression that gives you your > number of "phone calls" on a T1. > > As per -- > http://www.vocal.com/data_sheets/full/code_source_voip_g723.html > > Calls per T1 | Codec explanation > 289 or 240 |*G.723 (often referred to as G.723.1) - 5 1/3k > and 6.4k bps ACELP/MP-MLQ > 193 |*G.729 - 8k bps CS-ACELP *G.729A - reduced > complexity version of G.729 - fewer MIPS > at the expense of reduced perceived signal quality > 118 |*GSM 06.10 - 13k bps RPE-LTP > 96 |*G.728 - 16k bps LD-CELP > 96 to 38 |*G.726 - 16k, 24k, 32k and 40k bps ADPCM - > normally not used in Voice-over-IP applications > 48 |*G.721 - 32k bps ADPCM - normally not used in > Voice-over-IP applications > 24 |*G.711 - 64k bps PCM (A-Law or m-Law format) > > The above table shows one of the reasons G.729 is popular in > that you can get 192 calls per T1 with fair quality. > > Remember time is money; the tighter the compression the more > time it takes to compress/decompress and therefore the more > money in silicon it takes to do the compressions on the fly. > Smaller call "channel/bandwidth" means more hardware > horsepower to compress and decompress the voice on the call. > > Race "The Tyrant" Van der Decken > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Ed Greenberg > Sent: 16 December 2004 12:45 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Calculating required bandwidth > > I was posed this question: > > A T1 set up for voice carries 24 conversations on a circuit > that is 1.544 megabits/second. Right? > > Well, if you set that T1 up to carry data and run a link > between two IP networks over it, how many SIP conversations > could it be expected to carry? > How about IAX? > > How would one extend this calculation to varying bandwidth > circuits and various VOIP protocols (MGCP, SCCP and H323 come > to mind)? > > Rather than asking for a full education here, can somebody > point me at a suitable practical reference? Of course, if > somebody wants to actually post the answer that'd be fine too :) > > THanks, > </edg> > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
--On Thursday, December 16, 2004 9:45 AM -0800 Ed Greenberg <edg@greenberg.org> wrote:> I was posed this question: >I've learned a ton, in the discussion that followed this question. Thanks, all.
asterisk-users-bounces@lists.digium.com wrote:> I was posed this question: > > A T1 set up for voice carries 24 conversations on a circuit that is > 1.544 megabits/second. Right?Yes and no. If the T1 is channelized, then yes. If it's a PRI circuit, then it has only 23 channels to carry voice, as the 24th channel is used for the D-channel (signalling channel). PRI is superior, because it offers far more flexible use of the circuit, and provides far more information (like CallerID).> Well, if you set that T1 up to carry data and run a link between two > IP networks over it, how many SIP conversations could it be expected > to carry? How about IAX?Interesting question. I'll tell you this, it won't have so much to do with the 24 channels as it will with how efficiently the circuit is used. When you run data on a T1, all of the pipe is treated as one big channel by the upper layers. The 24 timeslots are all still there, but the network doesn't have any knowledge of them.> How would one extend this calculation to varying bandwidth circuits > and various VOIP protocols (MGCP, SCCP and H323 come to mind)?Each network layer (think of the OSI model) will add overhead, so the calculation has to take into account how the data (in this case, the voice packets) is encapsulated at each layer. Of the protocols, IAX would probably utilize the circuit most efficiently, due to it's trunking. Naturally, the codec you use will be another key factor.> Rather than asking for a full education here, can somebody point me > at a suitable practical reference? Of course, if somebody wants to > actually post the answer that'd be fine too :)I've always found Newton's Telecom Dictionary to be a great reference. It's not too technical, packed with humour, and very comprehensive.
asterisk-users-bounces@lists.digium.com wrote:> asterisk-users-bounces@lists.digium.com wrote: >> I was posed this question: >> >> A T1 set up for voice carries 24 conversations on a circuit that is >> 1.544 megabits/second. Right? > > Yes and no. If the T1 is channelized, then yes. If it's a PRI > circuit, then it has only 23 channels to carry voice, as the 24th > channel is used for the D-channel (signalling channel).Only if you're in the US. We have 30 + 1 :-) -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540
Hi,> >> A T1 set up for voice carries 24 conversations on a circuit that is > >> 1.544 megabits/second. Right? > > > > Yes and no. If the T1 is channelized, then yes. If it's a PRI > > circuit, then it has only 23 channels to carry voice, as the 24th > > channel is used for the D-channel (signalling channel). > > Only if you're in the US. We have 30 + 1 :-)Are you sure? As far as I know, E1 is 30 + 2, not 1... Best Regards, -- Durval Menezes (durval AT tmp DOT com DOT br, http://www.tmp.com.br/)