- This topic has 17 replies, 1 voice, and was last updated 20 years, 7 months ago by Compression.
25th May 2002 at 19:52 #21660SamatarGuest
Can anyone tell me with certainty the maximum simultaneous VOIP connections (calls)that I can get from T1(1.544KBPS) data (internet line). I am setting up a VOIP calling card business and my customers will be calling from thier homes using their PSTN phones. I am aware of that a T1 voice provides 24 phone lines of each 64KBPS but doesn’t a VOIP call take much less bandwidth than 64KBPS. Or each PSTN voice call coming from the PSTN has to be matched with a 64KBPS data line. Can I have a 2 T1 voice lines from the PSTN (48 calls)attaching to my AS5300 on one side and T1 data line attached to my router on the other side and then still get a 48 calls made. simultaneously.
I would really appreciate an aswer for this
Thanks25th May 2002 at 21:33 #21661RommelGuest
Answer is it depends…what codec are you using on the ip leg, what is your sampling period.
I doubt you will get 48 calls made through a single T1, I havent done the math. But you probably need a high compression codec like G.723.1 5.3k or 6.3k and cRTP enabled.26th May 2002 at 02:51 #21662SamatarGuest
Thanks Rommel for the answer. I am not sure which codec I will be using because I am not the one who will be configuring the router. My plan is to pay someone for that part of the work after I do everything else such as the phyical interconnectivity myself. Let us assume that I don’t want to compromise for the quality of the voice. Could you please then suggest a codec, the best one for quality of the voice and then let me know how many voice lines I can get from my T1 data line.
Thanks28th May 2002 at 11:29 #21663MasoodGuest
Give up Cisco. RAD can manage per CH 4.8 Kbps. So.. In T1 you can have 24 telephone chennels.29th May 2002 at 01:00 #21664RommelGuest
I am not sure if you are directing that Cisco comment to me, but I will have to admit that most of my experience lies with their voip products.
It is great to hear that RAD can squeeze 24 channels per T1 and would love to hear more about the product. Does the number you gave above rely on VAD efficiencies? What type of codec is it using? Can you send me a pointer?29th May 2002 at 03:44 #21665xavierGuest
I have been using G723.1 as a voice codec and had 8 fxo/fxs ports (8 calls at the same time)working on a 128 kbps internet connection, and had a prepaid calling cards platform running on it.
There are several choices in the market for gateways, i think most give you the option to choose from different voice codecs to work with.
Good luck29th May 2002 at 03:59 #21666SamatarGuest
Shouldn’t it be normal to get a 24 voice lines from T1. What is the surprise here.29th May 2002 at 18:16 #21667RommelGuest
Okay…I did not mean to say it was not possible. It is a trade off, voice quality vs bw consumption.
If you want complete toll quality voice and use a G711 codec you will not be able to do this. But if you are okay with poor quality voice and want to go with a G723 codec so be it, you will be able to pump more calls through it.30th May 2002 at 03:13 #21668SamatarGuest
I had been wondering which codec should one use to get the best quality VOIP if bandwidth was not an issue. Are you saying G711 codec gives you the best quality?30th May 2002 at 06:16 #21669zafar al masoodGuest
visit the site:http://www.rad.com/products/family/km-2000/km-2000.htm30th May 2002 at 22:02 #21670RommelGuest
For all intensive purposes, G711 gives the best quality with 10 ms packetization period is the best.
G711 samples a 0Hz to 4kHz frequency range. If you dont go with a standard codec you can probably get better quality out of a platform. Traditionally proprietary codecs work great, but then it wont work with other folks voip endpoints if they dont know how to code or decode those codecs.
There are some wideband codecs out there that will sample a wider range but I havent seen any implementations of those especially since they generally use higher DSP MIPS. Those will give better than toll quality voice.
Lastly, I have been hearing increasing interest in the GIPS(Global IP Sound) codecs. It seems as if they have enhanced many traditional codecs to be used for Voice over IP. If you know, or find a vendor that uses their codec they may have a lower bit rate codec with better quality voice under adverse conditions. This still presents the same issues as above…interoperability30th May 2002 at 22:11 #21671RommelGuest
Thanks for the link…I have a couple of questions:
G723.1 MPMLQ says its a 6.4k codec in the doc you pointed to…In ITU G723.1 it states its a 6.3k codec.
Why is there a difference?
PCELP…is that a RAD only codec? What does the P stand for?
It also looks like this is for point to point applications, is this extensible to a routed telephony environment i.e. any to any rather than a to b?
Thanks!!!31st May 2002 at 00:16 #21672Wilson BoyrieGuest
I am confuse now.The way that I understood the question is:
1)Customer call into the Cisco 5300 using PSTN lines,carried over T1 circuits provided by your friendly phone company.
2)Those T1 lines are connected to the T1 ports of the Cisco 5300.
3)Calls are answered by the Cisco box, customer get a prompt, enter account number and pin #, dials the remote number.
4) Call is packetized by the Cisco, converted to VOIP format and sent to the other end directly or via a ITSP (Internet Telephony Service Provider).
On this configuration the call enter the Cisco box on normal telephony format (64 KB per call, one channel per user, 23 or 24 calls per T1 depending on signaling type) and come out of the box over the ethernet port as packets of data. The bandwith used will depend on the codex used, but a rule of thumb will be around 12-15 kb of bandwith for each call if you want excellent quality.
If you want to cut corners and offer low quality you could get away with as low as 6 kb per call.
The answer to the original question is : You could get up to four T1 of traffic from the phone company on a full T1 (1.544 mb) of data on your ethernet port with very good quality.
That is a four to one ratio. If you go cheap you could do up to eight to one.
I have a woorking gateway (Quintum) that is running like a champ with four channels on a 64 kb circuit.31st May 2002 at 03:19 #21673Zafar al masoodGuest
For data rates of 6.4 kbps and above, voice compression is based on the MPMLQ (Multipulse Maximum Likelihood Quantization) speech coding technique, as defined in the ITU-T G.723 Standard. The compression at 4.8 kbps, uses a proprietary compression technique.
masood31st May 2002 at 23:05 #21674RommelGuest
I understand what MPMLQ is and in ITU G723.1 it states that this is a 6.3kbps codec. I havent seen an addendum so I just wanted to understand.
William- I guess it depends on what you call “very good quality” G.729 is close enough to toll quality…so I guess I would call it near toll quality but not _very_ good quality. lets face it if you play music to your friend over a land line the music just doesnt have the fidelity of having something like dolby theatre sound in your house. Thats my benchmark for _very_ good quality. (you can hear a pin drop….on a plush carpet) aaahhhh…the wonders of the ear. I believe the question was the _best_ voice quality over voip. The best would be the least amount of latency introduced and smaller packetization periods so a lost packet is not noticed. 🙂 right?
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