Do Any VOIP Clients Actually Work?

Lets say we set up an Asterisk server then people want to use it from home or on the road or somewhere to make calls, they will need VOIP client software to use to call with, right? Is there any VOIP client software that can call a REAL phone and ACTUALLY WORKS or am I wasting my time looking?
We’ve been trying to find some kind of working client software for quite some time but had no luck so far. You’d think that since anyone can do voice chat with AIM or Yahoo Messenger that something would be available but I can’t find anything that works. I tried Net-2-Phone a while back, started with a $10.00 credit and quality was so bad with lots of echo, delay and so on it was worthless so only attempted to make a couple of calls but gave up and still have $9.00 credit that will never be used. I have wideband access over a cable system so it should work. Also tried Skype with no success. I could hear callers but they couldn’t hear me. It works fine between 2 PC’s on our LAN but not over the internet. Skype has no support or help, just a problem reporting form that I never heard anything back from so problems can never be fixed. This would likely require someone inside with access to their software and servers but all they have are other frustrated users in the forum that try to help but just end up giving the same stupid advice over and over like “check the microphone”!
Lastly we tried iPhone. Roman from iPhone was nice enough to give me a test account and help with some setup problems like telling me which of many ports to open through the firewall. They use SJ Phone which looks a lot like Net-2-Phone without the ads. The text and illustrations in Help don’t actually match the program so getting it setup required some guessing and it has hardly any options or a directory or anything like that to find someone to call. You have to get the number from the internet or somewhere in order to dial anyone. I don’t know anyone else who has iPhone and it seems unable to call any phones outside of the iPhone system so I haven’t been able to talk to anyone using SJPhone except Roman when we were first configuring it. After telling a friend about all these problems he asked why companies keep producing phone calling programs that don’t work but I don’t have any idea. I am hoping someone does come out with something that works but I haven’t been able to find it so what do you folks use with Asterisk, or do you even try to do this at all? Thanks.

I am not understanding your question, but I can give you the following idea of my system:

  1. Dual Asterisk Servers running behind a LINKSYS Router/Firewall

  2. Xten EyeBeam softphones on some PC’s.

  3. SIP Hardphones on some desk.

Now the good news, is I have 2 Analog lines out to the PSTN Network.

My family in Germany can call anyone here in the USA, they can just pickup there SIP Phone in Germany and ring our extensions here in the USA, in Texas, Lousiana, Illinois and such… Best part of all of this all calls use the Asterisk box and all calls are bascilly free long distance… The calls that leave over PSTN are on SBC and I pay the flat long distance rate of $20.00 per month… QUALITY… I can hear a PIN DROP…

Now when any one is in a Hotel with DSL or what, they can call on there xten eyebeam softphones and if they have video everyone can see as well…

Now also if no-one is at there exten… then it just goes to voice mail… then it sends out an email and sms message to there cell phone…

Now does it work… I think so… :smile:

Then you have been able to get Xten EyeBeam softphones to work successfully?
That brings up some more questions:

  1. How can Asterisk Servers run behind a LINKSYS Router?
    Did you disable NAT and open every port? From what I’ve read it would seem nearly impossible if you are using NAT.

  2. If Xten EyeBeam softphones use UDP they can have a lot of packet loss so how do you get good quality audio in spite of all the delay and echos?

  3. Do Hardphones work any better than Softphones, or does it matter?

I am at work right now, but here is a quick update, I will reply more later.

  1. I use the Linksys WRT54GS as my connection at home to the internt which I have SBC DSL, with a dynamic IP.

  2. I use as my Dynamic DNS Host.

  3. I use DirectUpdate to keep my IP Updated, with www, and a few A records for my machines.

  4. On the Linksys box, I have port forward setup to forward PORT 80, 5038, and some 7000 range IP’s which I can’t remeber to my one Asterisk Server with an internal IP of say 192.168.1.XXX or what ever you want.

  5. On the Xten phones, I use the domain name as my Proxy and such so where ever there is a place for network, etc., etc., etc. proxy I use the Domain Name I have at Zoneedit… like say or what ever… This is on the Xten Phones.

  6. On the phones that are on the local Internal network, I just use the 192.168.1.XXX of the Asterisk server as their IP Proxy server, etc., etc., etc…

  7. All of the Softphones and Hard phones are setup in SIP as an phone… I will post this when I get home…

And thats about all I do… Sounds great, works great, Video is great…

This way you don’t need to have a registered IP, and I use all registered products, I don’t use DEMO’s…

P.S. My two ASTERISK BOXES are ----- DELL OPTIPLEX GX100 with 512 RAM, 20GIG HARD DRIVE (ebay around $60.00), Mandrake Linux 10.0, and the 2 Digium Developer Kits… I have a total of about 20 phones on right now…

Now for some really cool features, I bought 2 Linksys Ethernet Bridges on ebay, and 2 IP hard Phones… I then plug the computer into the PHONE, the phone into the Linksys Ethernet Bridge, and now you have a WIRELESS PHONE and COMPUTER for about $90.00 a far cry from the $200.00 and up price tag of others… The IP Phones configure so easy via a WEB Browser…

My system has the folowing, Directory Dial, VoiceMail, Streaming on Hold Music (staright from the Internt, no more pre-recorded music, a little tricky), WEB Mail, WEB Call, MySql CDR, and on and on and on… :smiley:

As far as VOIP clients that work goes, yes - there are lots of them!

SJPhone works. For example, I was using it last night to talk to my mum, who’s on the other side of the world. It was running on my laptop and connecting to my asterisk server, which is a linux box with a netfilter firewall running, on a dynamic ip address ADSL connection. I phoned her on her home landline phone via a SIP connection to

I’ve used a number of SIP and IAX2 softphones and they’ve (more or less) all worked. Some work better than others and some have features i prefer to others etc. Getting the whole VOIP setup to work right, though, is another matter altogether.

And VOIP service providers are yet another matter again! I’ve got accounts with 5 of them at the moment - 2 in Australia and 3 in Europe - and the service and PSTN termination quality definitely varies.

Read for asterisk-specific information.

Now for some really cool features, I bought 2 Linksys Ethernet Bridges >on ebay, and 2 IP hard Phones…
Thanks for all the info.
What kind of IP hard phones can you get cheap?

I bought some AriaVoice Phones…

Work great, has WEB Interface… Love’em…

Why do you think the hard phones work so good? I don’t understand how they could be better than a PC running a phone client but I guess it depends on how good the software is.
I tired SJ Phone and it was TERRIBLE! That’s why I was asking about the software. I’ve only been able to make 2 calls but the audio quality was so bad it wasn’t usable. The audio was very weak with lots of noise, echos and delay which makes me think that VOIP is not ready for prime time yet.

Try an iax phone, it will give you less problems with nat firewalls at those locations.

idefisk, is such a phone.
You could find many more (diax, iaxcomm, … with google)

Thanks, I’ll try that.

Are you using SJPhone on windows or linux? I’m using the linux version and i haven’t had any problems with the audio quality at all - not related to SJPhone, anyway.

All my audio quality problems (and very likely at least some of yours) seem to be caused by a cheap and nasty ethernet switch and various other hardware issues. SJPhone, if anything, works better, audio-wise, than iaxComm (an iax2 softphone).

The other area where audio quality problems come into the picture is your internet connection. If you’re using an ordinary ADSL connection, you’re unlikely to get top voice quality. Depending on your connection speed and your service provider and what sort of contention ratio they work with, your experience might be better or worse.

Some of my audio problems show up in the connection between my laptop and my asterisk server - which are connected by a 100bps LAN. These problems are nothing to do with my internet connection. However, i get other problems which are related to my internet connection. Add these together and it just gets worse.

One of the main problems of ADSL connections is the low upstream bandwidth. So you may hear the other end ok, but your voice sounds appalling to them.

I’m trying to use SJPhone in Windows XP Pro and I’m on broadband over a cable system. Download speed is 3Mb but upload is only 128Kb. I have no idea what kind of ethernet switches the cable co. has.

I’ve just downloaded and installed Idefisk and tried to configure it with no luck.
There does not seem to be enough Account Options to enter all the necessary settings.
Besides my name, number and password I also need to enter SIP Proxy of and port 5060 but where do I put those settings? I tried these in the Server box but it would not Register and gives a Status message saying non-existant sesson, Dropping!
Also it won’t save the settings so when I start it again it says Config File Not Found! and the Account Options are all blank.

I also tried IAX Phone from Sokol & Associates and that looks a lot nicer but it does not work at all, the part of the instructions telling how to configure it are not finished!
Both of those programs are still in beta. All the options and settings for IAX Phone are different from Idefisk which are all different from SJPhone!

Doesn’t anyone in this business ever finish anything?
If they gave up because they didn’t have time to finish it and couldn’t get it working why don’t they take it down so people won’t get frustrated trying to use it?

[quote=“stevec5000”]I’m trying to use SJPhone in Windows XP Pro and I’m on broadband over a cable system. Download speed is 3Mb but upload is only 128Kb. I have no idea what kind of ethernet switches the cable co. has.

The ethernet switch is nothing to do with the cable company. I was talking about my local network, but if your windows computer is connected direct to the cable network and you haven’t got a LAN, it’s got nothing to do with your setup.

A 128kbps uplink should give you sufficient bandwidth for ip telephony. However, that depends on what codec you’re using (in the audio/compression settings menu in SJPhone, i think - it is in the linux version, anyway).

It also depends on whether or not you’re really getting 128kbps to yourself. All ISPs have what’s called a “contention ratio” - which is how many people have to share how much bandwidth.

They all work on the basis that all subscribers on a particular circuit aren’t going to be using all their bandwidth at the same time, so they don’t need to provide 128kbps for each person - maybe they’ll have that much for every 5 people, or every 10.

So if their contention ratio is high, you’re likely to have problems. Also, if you’re trying to use it at peak usage time, you’re likely to have problems.

Using a lower bandwidth codec (e.g. gsm) - if possible - may improve matters.

There are other factors that influence the sound quality you’ll get. The cable modem you’re connecting through is likely to be one of them. As is you’re computer and how it’s working (although that’s probably the least of your worries).

It seems that I can’t use Idefisk or any sort of IAX program (even if they were finished and had setup instructions) because I don’t have access to the server, just a user account, and can’t configure the necessary codecs and other perameters it needs at the server end.

If you haven’t got an iax account, you can’t use an iax softphone. If you’ve got a sip account, you’ll have to use a sip softphone (or hardphone, of course).

You don’t need access to the server to set up what codec you’re using. You do it in the phone’s configuration. However, you can only use one of the codecs that the service provider’s server supports.

Some service providers make some available, others make others available. You have to find out what codecs your service provider supports and find out what your softphone supports and choose one or more of those codecs you’ve both got in common.

Everyone always seems to support ulaw - but that’s a high bandwidth codec. GSM doesn’t seem to be so widely supported by ITSPs, but if it is, it’s generally a good choice, as it’s low bandwidth, but decent quality. ILBC is sometimes available too, and that’s low bandwidth as well. G729 is a low bandwidth/high quality codec, but it’s proprietary and it costs money to use it. It’s not included in SJPhone, but some ITSPs support it.

I had the SJPhone client working over the weekend and made a couple of calls ok but this morning it won’t work at all! It took quite a while just to register then when I try to call it just gives me an error!
I really can’t see why everyone is so excited about something that isn’t going to work when the internet is busy? When all the packets are being dropped nobody is going to be able to make any calls so it seems futile, nothing but a big waste of time.
What’s the point of installing phones systems that will never be reliable?

Asterisk and VoIP in general is relatively young technology.

It will never be the answer to everything with regards to telecommunications. But, for what it does, and will do, it is great. You just have to know its strengths and weaknesses.

If the weaknesses are not being able to make any calls on Monday morning or most of the rest of the week what are the strengths, free calls on weekends? Since it depends on UDP packets that get dropped, how can it ever be improved? Is someone going to reprogram all the routers in the world to give UDP priority over everything else so the free calls can go through? I guess it’s possible but I don’t ever see it happening. The only way this will ever work with reliability is when a new kind of packet is invented for telecommunications that has different characteristics and can’t be dropped.

Not UDP, no. But QoS is inevitably going to become more important and will probably be implemented widely.

The only way you can have a packet that “can’t be dropped” is by making it analogue and sending it down its own dedicated wire! Kinda like a telephone line, really…


Saying all this VOIP software is crap because of poor audio quality is quite a big incorrect sweeping statement. I’ve been using Skype (Also asterisk over vpn connections) for months now to call all over the world, pc to pc and pc to landline and the audio is crystal clear. I’ll add that I am on a standard 512kbs adsl connection in the UK (which is well known for it’s slow internet).

I think you should look a little closer to home and find out what’s wrong with your case.

Do you have any other computers in your house you can try skype/voip softphones on? If it’s still poor quality I strongly believe that your internet connection is the source of your problems.

(PS Slating Skype support for a free software product? I’m sure if you offered to send them your system, or a system identical to yours so they could reproduce the error they’d do it! - Crazy - Just how many unique support requests do you think go unanswered by Microsoft every single day?)