Voice over wirlesss

Paparattzi

Hi,

Just wondering if anyone has experience of telephone service over a wireless connection such as Irish Broadband Breeze or Digiweb Metro services ? this is a general technical question, as I am going to talk to some of the operators and would like to hear some real experience before I do.

According to my estimates these service have enough bandwidth at 3 or 4Mbit/s and the latency looks ok, but I wonder if the connection is stable enough for heavy duty voice use?

Paparattzi

SkepticOne

Paparattzi wrote:

Just wondering if anyone has experience of telephone service over a wireless connection such as Irish Broadband Breeze or Digiweb Metro services ? this is a general technical question, as I am going to talk to some of the operators and would like to hear some real experience before I do.

According to my estimates these service have enough bandwidth at 3 or 4Mbit/s and the latency looks ok, but I wonder if the connection is stable enough for heavy duty voice use?

Bandwidth requirements are not high. You don't need anywhere near 3-4 Mbit/s. How many lines are you talking about? Main concern is reliability of broadband connection.

Note that Digiweb metro is not a typical VoIP service but uses similar techniques that cable companies use to provide phone services. Different channels are used for voice and data. Packets do not go out over the public internet.

If you have a reliable broadband connection and a good VoIP provider then whether it is wireless or whatever should make little difference.

Try the VoIP forum maybe for technical answers.

Blaster99

I've used VoIP over Clearwire and Irish Broadband. Both were poor. Any sign of packet loss and VoIP is dead as disco.

cgarvey

Moved IoffL > VoIP

Paparattzi

I estimate that each voice line required about 300 to 500kbit/s for good quality. Since this is both upstream and downstream, this can be a problem with ADSL.

Paparattzi

bhickey

Paparattzi wrote:

I estimate that each voice line required about 300 to 500kbit/s for good quality. Since this is both upstream and downstream, this can be a problem with ADSL.

Bandwidth required depends on the codec that you use. G711 will need about 80kbits/sec each way whereas G729 needs about 30kbits/sec.

Paparattzi

bhickey wrote:

Bandwidth required depends on the codec that you use. G711 will need about 80kbits/sec each way whereas G729 needs about 30kbits/sec.

This is perfectly true, but don't forget we are talking delivering vocie over a packet switching netwwork, not a synchronous network, so the newtwork bandwidth requirements are much larger.

Paparattzi

bhickey

Paparattzi wrote:

This is perfectly true, but don't forget we are talking delivering vocie over a packet switching netwwork, not a synchronous network, so the newtwork bandwidth requirements are much larger.

Please elaborate. The bandwidth figures I gave represent traffic measured at the router. If I have a standard Eircom Business Starter broadband ADSL package at 3Mbit/384kbit then there's plenty of room for several simultaneous VoIP calls. I've done this and it works perfectly.

Paparattzi

bhickey wrote:

Please elaborate. The bandwidth figures I gave represent traffic measured at the router. If I have a standard Eircom Business Starter broadband ADSL package at 3Mbit/384kbit then there's plenty of room for several simultaneous VoIP calls. I've done this and it works perfectly.

When you say ther is plently of room for several VOIP calls, have you actually tried this, with ordiany internt traffic also running ? My estiamte is that 384kbit/s is ok for one VOIP line, using a best effort IP service. however if you have MPLS you could do a bit better, but your router would need to be set up to handle the tags and you would have to lower ther priority of other traffic types, even SAP traffic which could casue some grief.

Paparattzi.

bhickey

Paparattzi wrote:

When you say ther is plently of room for several VOIP calls, have you actually tried this, with ordiany internt traffic also running ?

All day every day in several sites. Any home router with QOS (Quality Of Service) capability should be fine for several simultaneous G729 calls unless there's some pretty extreme Internet traffic going on. Obviously a separate dedicated ADSL line would be nice but for most small offices it's not necessary.

SkepticOne

Paparattzi wrote:

My estiamte is that 384kbit/s is ok for one VOIP line, using a best effort IP service.

How is this figure arrived at in your estimations?

Paparattzi

SkepticOne wrote:

How is this figure arrived at in your estimations?

My estimate is that is takes about 240kbit/s for a voice call over a best effort IP service, taking account of latency and trying to keep packet loss less than 1%, to achieve voice quality near PSTN or ISDN quality. you have to take account of self-similarity of IP traffic, so the calculation is complex. It assumes a demand rate of 80kbit/s and a maximum traffic loading of 33%.

Paparattzi

fisab

Let me simplfy the situation.
1 user, 1 broadband connection, 1 router, 1 ATA. 1 outgoing phone conversation.
Suppose I connect at G711 (80kbits/s).
Surely I need only 80kbits uplink plus some overhead to talk?? Am I wrong?

If both parties talk at the same time do I need 160kbits plus overhead to talk without a degrade in quality?

aaronc

fisab wrote:

Let me simplfy the situation.
1 user, 1 broadband connection, 1 router, 1 ATA. 1 outgoing phone conversation.
Suppose I connect at G711 (80kbits/s).
Surely I need only 80kbits uplink plus some overhead to talk?? Am I wrong?

If both parties talk at the same time do I need 160kbits plus overhead to talk without a degrade in quality?

No you only need 80Kbps* in either direction so 80Kbps of your downstream link and 80Kbps of your upstream link.

What Paparattzi may be alluding to is that in order to get a consistent 80Kbps connection you generally have to purchase a product quoted at 4 times the capacity required. This would not be a perfect way to do things but it's probably not a bad rule of thumb. Every ISP is different and even for the same ISP each location is different. This is particularly so with DSL connections where one exchange can be running over capacity and another one can be running under capacity. The same quoted DSL product on different exchanges can be vastly different. For example bhickey would not be so lucky with his simultaneous calls if his connection was in Dublin 2 .

* g711 at 20ms packet size = 160B
RTP Header + UDP Header + IP Header = 12B + 8B + 20B = 40B**
Total = 200B/packet
Transmission rate = 200B packets x 50pps = 10,000Bps = 80Kbps.

There will be additional overhead for the Data Link Layer protocol but the transmission rate quoted by your ISP should be for the Network Layer so that can be ignored.

** Can vary slightly and the link below quotes g711@20ms as requiring 82.8Kbps of bandwidth.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml

Aaron

bhickey

aaronc wrote:

The same quoted DSL product on different exchanges can be vastly different. For example bhickey would not be so lucky with his simultaneous calls if his connection was in Dublin 2

Luckily for me I live in the shticks and I'm the only person in my town that uses the Internet during the day.

sirlinux

Lets throw in full duplex into the equation as well, try running any protocol that has an rtp stream, voip especially, or video over a half duplex link and it's not going to work.
An el cheapo wireless connection is not going to have the two radios need to transmit and recieve at the same time, a jitterbuffer will only do some magic for you.

Paparattzi

aaronc wrote:

No you only need 80Kbps* in either direction so 80Kbps of your downstream link and 80Kbps of your upstream link.

What Paparattzi may be alluding to is that in order to get a consistent 80Kbps connection you generally have to purchase a product quoted at 4 times the capacity required. This would not be a perfect way to do things but it's probably not a bad rule of thumb. Every ISP is different and even for the same ISP each location is different. This is particularly so with DSL connections where one exchange can be running over capacity and another one can be running under capacity. The same quoted DSL product on different exchanges can be vastly different. For example bhickey would not be so lucky with his simultaneous calls if his connection was in Dublin 2 .

* g711 at 20ms packet size = 160B
RTP Header + UDP Header + IP Header = 12B + 8B + 20B = 40B**
Total = 200B/packet
Transmission rate = 200B packets x 50pps = 10,000Bps = 80Kbps.

There will be additional overhead for the Data Link Layer protocol but the transmission rate quoted by your ISP should be for the Network Layer so that can be ignored.

** Can vary slightly and the link below quotes g711@20ms as requiring 82.8Kbps of bandwidth.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml

Aaron

Thanks for the article reference from Cisco, I haven’t seen it before. I just glanced over it, but it does not address the question we are discussing. It concludes that you generate a 82kbit/s payload for a G.711 voice call. I was using 80kbit/s so not too different.

But note that Cisco is very careful to call it payload, and not network bandwidth or network capacity, this because the bandwidth you actually need is very network dependent.

To calculate the bandwidth you need to take account of the packet loss on the type of network you are using, and this depends on, amongst a lot of things, the traffic load you are sending into the network. IP traffic is inherently bursty, even traffic like voice is sent in bursts, and this causes packet loss and this in turn cause voice quality degradation. to avoid this you need a lot of bandwidth overhead. This Cisco article does not address this, but others article looking at network capacity do cover this issue and recommend no more than 30-40% loading of a VPN link. The overhead needed for a crappy DSL line is not stated. I will try track down an example article later.

Ok you can argue that a VOIP service is inherently poor quality, so there is no point in trying to make it better. But my feeling is that if state of the art technology can provide better quality than telephony which was invented 150years ago and perfected 40 years ago, we should forget the whole thing.


Paparattzi

Blaster99

IP networks are not designed to carry voice and VoIP codecs are not designed to deal with packet networks, so you're not exactly looking at a marriage made in heaven.

Darth Maul

I can't get broadband in my house "line fail" so I got it in a house 2 miles away and setup a wireless link, basically I have a wifi ADSL router in the house with the ADSL, attached a 9dbi directional aerial, the on my house I have a 24dbi grid aerial connected to a wireless bridge, only signed up fo BT 1mb service until I'm sure I'm getting a reliable connection, and so far after a month, Its great, Have a VOIP ATA connected and so far its been great, good quality,
tried a few differnet codecs but found G726-32 has been the best, G729 was far less stable even though it has much lower bandwidth requirement. At tme moment I'm using Voipstunt,
only problem I found is if someone is using the internet at the same time as the VOIP the quality drops considerably, but this is understandable due to the codec I'm using and the only having a 1mb service.

kroc

forgive my ignorance but, what are all these codecs (e.g. G726-32 ) you are talking about? Is it a router thing ?

Darth Maul

kroc wrote:

forgive my ignorance but, what are all these codecs (e.g. G726-32 ) you are talking about? Is it a router thing ?

A codec (Coder/Decoder) converts analog signals to a digital bitstream, and another identical codec at the far end of the communication converts the digital bitstream back into an analog signal.

In the VoIP world, codec's are used to encode voice for transmission across IP networks.

Codecs generally provide a compression capability to save network bandwidth. Some codecs also support silence suppression, where silence is not encoded or transmitted.
Codecs vary in the sound quality, the bandwidth required, the computational requirements, etc.

Each gateway or softphone, typically supports several different codecs, and when talking to each other, negotiate which codec they will use.

Blaster99

G.729 is very sensitive to packet loss, that's why you're seeing problems on an unstable connection. I tend to use G.711, which is marginally better in that respect.

bhickey

The iLBC codec is described as "a low-bit rate, narrowband codec with high packet loss robustness". A lot of ATA's and IP phones don't support it though. I've tried it with a Grandstream 486 but went back to G729 again which still works best for me. Best thing is probably to read up on the codecs and then try out whatever ones you think will suit you best. There's loads of info on VoIP codecs at :


http://www.voip-info.org/wiki/view/Codecs