I tried many VOIP phone services, and found that some will provide you with good quality phone service, especially like sound is clear, call is smooth and not stuck sometimes. So except the internet connections, which part of the hardware and software affect the VOIP phone quality?

The type of codec used can play a factor in quality (g.711 vs g.729) on the software side. As for the hardware side, some dedicated voip phones (polycom ip650's, 670's, etc) offer things like echo cancellation, noise reductions and other "tricks" to make the call sound clearer.

Sorry, I'm not going to use gateways. I want to setup a similar voip service like skype. So for the hardware side, only a server is required. I want to know for this kind of voip service, what're the main part affect the phone quality.

We have several VOIP providers on our network, and the biggest concern is always latency. Low latency = good call quality. Single hop's to end client networks will provide you with a good solution. Look to colocate your server with a multi-homed internet provider!

Don't forget you should also have a sip trunking provider who has a good quality of routes. Another thing cheapest rate for the call does not mean a good quality of the call. And the biggest problem of quality is Jitter. If have a latency over 200 ms even with packet loss of 10% it still should not effect your voice quality as long is you ITSP has a good SBC (Genband or Acme).

Server load also plays a big part in call quality. If the VoIP switch (server) has a heavy load, it can cause quality problems. Your upstream provider plays a role in that...overall, latency is going to be what to watch for in terms of good or bad call quality. If you have high latency or a lot of hops to get to the destination, it could cause poor quality issues.

In addition to what has been discussed above. Don't use a VPS for VOIP unless you know how to tune it. Don't mix codecs. Use Ulaw. If latency is an issue, configure a good jitter buffer. Using above details, I have been able to get a quality VOIP configuration using SIP and 1300ms - >2000ms over a satellite connection between San Francisco to Iraq.

network latency

I've found the following site useful in troubleshooting my VOIP connection when it goes awry:

http://myvoipspeed.visualware.com/


Our data center always gets a score of "A" in the testing process, but my home cable modem connection can fluctuate quite a bit. One nice thing about the Visualware site is that they use multiple testing sites throughout the world. Because of this, you'll be able to see if poor call quality and/or a low score in the test process is related to routes or not.

I just bought a DELL server, and will buy the softswitch system and softphone soon. Above I heard many talking the latency. Is that a configuration in the softswitch system?

Any suggestion for the softswitch system and softphone software?

Latency is the time it takes for a package to travel between the phone and the server. It is usually linked to the physical distance and the number of hops.
If you look at transatlantic latencies, count on 150ms tot 250 ms. Latencies within Europe are limited to 50 ms.

Your server should be as close to the user as possible.

a lot of people here mentioned latency and jitter, etc. but did not really explain how this can affect your voice service.

Yes latency can affect your service depending on how you are planning on terminating your calls. VoIP has two components, a signaling component and a media component. Both component doesnt necessarily have to go thru your network. Media or RTP can be sent directly from the originating UA to the terminating UA - completely bypassing your network. How does this affect your service quality? Well in this case the metrics measured on your network goes out the window - your trans-US latency of 25ms is completely irrelevant if your customer has 500ms latency with 20% packet loss to your terminating carrier. One way to completely control this is by using an RTP proxy to insure all media traffic goes thru your network(this also helps issue with 1-way audio, nat traversal, etc). The list can go on and on..

As you can see - its easy to set up a voip service but creating a carrier grade voip service has more complexities to it that just ordering the fastest,bad-ass internet provider out there.

Thanks all for the answers. Any suggestions for cheap and good quality softswitch then?

I think "cheap" and "good quality" can not be combined together. You should pick one or the other. If you want a recommendations for application you should specify your needs.

Low bandwidth?

All those things, latency/jitter are definitely the related to VoIP quality issues, but a lot of the time those are side effects of network issues and not issues in themselves.

For example, a good Firewall or Linux/BSD based Router will typically prioritize packet flow based on port, protocol, or tag. All it takes is one person running Bittorrent to ruin a VoIP setup because it's very demanding (in terms of packets-per-second) on hardware and it can't keep up.

Packet loss is the number one killer of voip performance which uses UDP and has no way to resend lost packets. the result is choppy or scratchy audio.

If you are communicative, easy-going and sociable person, if you like to discuss the latest events with friends, if you have so many friends all over the world, you need not expensive means of communication. We propose you the VoIP catalog owing which you can be closer to the friends. Just imagine the call duration and calculate the bill for talks.

Packet loss is the number one killer of voip performance which uses UDP and has no way to resend lost packets. the result is choppy or scratchy audio.


Very true packet loss is the number one killer in performance, but usually there are underlying issues causing is, that most likely can be resolved.

Packet Loss can occur for a variety of reasons including link failure, high levels of congestion that lead to buffer overflow in routers, Random Early Detection (RED), Ethernet problems, and the occasional misrouted packet

Packet Loss typically occurs in bursts of 20-30% loss lasting 1-3 seconds. This may mean that the average packet loss rate for a call appears low although the user reports call quality problems.

If packet loss is accompanied by a high level of jitter or packet loss is very bursty then the problem may be congestion related.

If packet loss is present when the jitter level is very low or if the problem is specifically associated with one or more users on a specific Ethernet switch then check for Ethernet problems.

If you are communicative, easy-going and sociable person, if you like to discuss the latest events with friends, if you have so many friends all over the world, you need not expensive means of communication. We propose you the VoIP catalog owing which you can be closer to the friends. Just imagine the call duration and calculate the bill for talks.

I'll give $5 to anyone that can explain to me what he just said.