- This topic has 12 replies, 1 voice, and was last updated 15 years, 9 months ago by CCD.
31st March 2005 at 13:07 #29275VOIP BeginnerGuest
Im getting a little confused with the bandwith requirements for a particular link.
If I want to calculate how much bandwith I will need to support 1 call ( 2 parties) how much bandwith do I need @ G.711?
Assuming 80kbps (64+16) does this mean for 1 conversation i require 2x 80 (160kbps).
Now if the above is correct, how does ISDN do 1 call bothway in 1 64k channel (and signalling in another) when using the same sample rate.4th April 2005 at 07:36 #29276FrankGuest
For VoIP with G.711, you generally need 64kbps + x in either direction (64+x (up) AND 64+x (down)). That’s because you have to build up 2 separate voice paths (one from subscriber A to subscriber B and another one in the opposite direction).
By the way: if you assume a usual ip network, “x” (in the above calculation) is NOT 16 (add up header sizes of rtp, udp, ip + (eventually) ethernet)! Or did you mean a plus of 16 kbps for signalling? usually you don’t have to count the signalling bandwidth for voip because signalling is done in a totally differenz way than it’s done in isdn. there’s no need for a certain signalling bandwidth. have you heard about the session initation protocol (SIP)? That’s just one possibility to do signalling in voip.
ISDN uses ONE voice path (one so called B-channel) for bidirectional talking, which means that the voice data from A and B are added up and transported in an overlaid manner in the same data stream. That’s a main difference between voip and isdn. in voip, you open up two unidirectional “voice channels” (one from A to B and one vice versa), in isdn you just have ONE channel for both directions.5th April 2005 at 11:41 #29277dmckGuest
Not quite correct Frank.
ISDN is a signalling protocol using an E1( 2Mb) or T1 (?Mb) link as the carrier. As such it has a send link and a receive link. At the B channel level this translates to 2 seperate 64kb channels per conversation, one for send (A>B) and one for receive (B>A)
It is really only at the LAN level that the send and receive packets are on the same transmission media and the bandwidth capacity per conversation MUST be doubled to calculate the b/w of the LAN.
Regardless of the technology used for transmision between nodes the transmission link usually consists of seperate send and receive channels. THe only exception I can think of is when the link is another 802.? connection, ie ethernet cable.5th April 2005 at 15:49 #29278FrankGuest
for E1/T1 trunk lines you are right. I just figured it out from the ISDN BRA subscriber line (Uk0) point of view. There you’ve got 2 B-Channels (which in fact are both bidirectionally used) with 64kbps each plus one 16kbps D-Channel for signaling.
That kind of bidirectional voice transmission over one 64 kbps channel is realized by the use of echo cancellation (ITU-T G.165/G.168 I think). Unfortunately I don’t know the english expression for that kind of technique that is used to transmit the voice stream bidirectional A B on just one 64 kbps B-channel (in german it’s called Gleichlageverfahren). But it really works that way, otherwise it would not be possible to have two independent calls on just one ISDN BRA subscriber line (which in fact is possible).
I just didn’t think about E1/T1 because VOIP Beginner wrote about (64+16) kbps. I think with the “+16” he means the bandwidth of the D-Channel that’s used for signaling on a BRA subscriber line.
Frank7th April 2005 at 06:02 #29279dmckGuest
I thought that he meant the 16kb I.P. overheads added to the 64kb voice packet.
Your thoughts on bi-directional data channel is interesting. I’ve always assumed there is a 144kb transmission in each direction (2B+D) but that they are operating at different carrier frequencies, so that they in effect become seperate send and receive paths, but operating over a single pair of wires and would require 188kb b/w (plus some seperation)7th April 2005 at 22:39 #29280FrankGuest
the overhead b/w (dimensioning: kbps) for voice data carried over IP can’t be calculated in general. Amongst others it depends on the codec used and on how many bytes of voice data you carry per packet (this number is not codec-specific, so it can vary even for the same codec).
The number of overhead bytes that are transported with each voice-carrying packet can be calculated if you know the header length of all protocols that are used. RTP header length is minimum 12 bytes, UDP header is 8 Bytes, and so on.
Either way, “16” is not a number that I would say is typical for voip overhead as far as I know.
I had a look for the ISDN standards in the last days but unfortunately I couldn’t find the right passages by the time that I could affort. In fact on copper-wire based ISDN BRA (and as far as I know even on PRA) subscriber lines that I know definetely no FDM is used to provide 2 independent full duplex B-channels. Each subscriber NT adds up his outgoing voice data on the wire and filters out (using echo cancellation) these data in real time to make sure the TE only sees the voice of the opposite subscriber. As far as I found it this method is not necessarily true for all countries, so there might be other solutions around the world.
Frank8th April 2005 at 11:29 #29281dmckGuest
It works at voice frequencies so there is no reason it wont work for BRA!
I thing most countries are accepting ETSI standards for ISDN, I know Australia (developed their own before the ETSI standards were complete. WE have now completely dropped the local BRA standard and the local PRA standard remains in use for older equipment where manufacturers will not develop/upgrade to ETSI.
Thanks Frank, an interesting discourse and I’ve learnt something new.8th April 2005 at 15:29 #29282FrankGuest
ITU-T G.961, Chapter 5, ECH, explains what I was talking about concerning ISDN BRA bidirectional transmission by overlay.
Frank9th April 2005 at 10:06 #29283dmckGuest
thanks Frank, I shall look it up.15th April 2005 at 09:00 #29284tassGuest
each call on voip requires 10 to 13 Kbits/s of bandwidth
i guess so15th April 2005 at 13:47 #29285FrankGuest
what codec do you refer to?
Frank5th May 2005 at 06:16 #29286AliGuest
Can you tel me you people, When we use ISDN for voice, Is thier any codec technique involved?16th May 2005 at 18:09 #29287CCDGuest
Hi! i have a similar doubt about VoIP.Right now i’m doin’ a ‘project’ about this issue and im quite confused with the bandwith requirements as well.Using a G.711(64Kbps) or G.729(8Kbps) codec, which will be the new bandwith including RTP/UDP/IP headers?? 80Kbps per each direction in G711 case???in G729???.
If i have to offers full VoIP service to 648 customers in the rush hour at the same time, what kind of gateway do i have to buy??How do i know de number of customer it can give service to?how many calls can it support??…thank u