Vonage VoIP Softphone for Smartphone

[Guest kam_]

you can put some of the CF2 files onto the storage card, but its a bit complicated.

Woize use their own proxies which connect to SIP servers, but there is no inherant SIP facility in the client.

Bottom line, you can't point it at your own SIP provider, and so you're tied into their rates.

Its very usefull for incoming calls tho, especially if you go abroad, you friends can call you on a local number while your in the hotel and using WiFi

[Guest flodis79]

Thanks! I downloaded that .cab and installed but when I open x-pro.exe in the phone it says 'Evaluation complete. Thank you for evaluating X-PRO. Please contact your vendor for more information.' Strange..

Is that the version you downloaded yourself? I extracted it with WinCE Cab Manager.

Instead, I found ineen.com - they use the X-PRO software - and their softphone for PPC, which I'm able to run, but I cannot call another user's internal Ineen number, even if both users are online. I don't know if I'm doing it correctly with using Line1 and Line2, but at least there is a trunk tone coming out of the speaker.

Have a look at http://www.ineen.com/download_ppc.html

Back to the X-PRO again.. I exported the registry value from the Win XP registry to the smartphone, but it's looking for proxy all the time.. I have ports 5060 and 8000 open for the wifi connection (through NAT), but it's still not working.. I can see in the display that it finds the sip information (voipbuster) because it displays my display name, but it's looking for the proxy all the time. I tried to disable the firewall totally for a while, but still no luck.

What can be wrong?

[Guest kam_]

is there a NAT option anyway? it may be called NAT proxy or something.

I had similar problems with my ATA186, i had to set sip.sipdiscount.com:5060 as the NAT proxy

[Guest flodis79]

is there a NAT option anyway? it may be called NAT proxy or something.

I had similar problems with my ATA186, i had to set sip.sipdiscount.com:5060 as the NAT proxy

I only have Static NAT where I can set the internal ip assigned to the SP by the router.

Then I can open ports manually. But no NAT proxy option.. Can we solve this in some

way? The voipbuster client on the PC works just fine, it's just on the SP via XPRO it's

no go...

[Guest ampz]

you can put some of the CF2 files onto the storage card, but its a bit complicated.

Woize use their own proxies which connect to SIP servers, but there is no inherant SIP facility in the client.

Bottom line, you can't point it at your own SIP provider, and so you're tied into their rates.

Its very usefull for incoming calls tho, especially if you go abroad, you friends can call you on a local number while your in the hotel and using WiFi

If I put thoose CF2 files on the storage card, how much space will I need in the built in memory?

I don't understand why Woize are using .net 2.0 anyway. Most other smartphone applications are happy with just a few hundred kilobytes of space on the storage card, and require no supersized libraries to be installed.

It is true what you say about woize smartphone and SIP, but they are adding SIP capability to some of their other woize clients (the symbian client for example), so perhaps they will add it to the smartphone version sometime in the future as well.

Their servers support SIP clients, so you can use any SIP client and call using their services.

It is unfortunate that Woize and Skype use proprietary protocols, but their protocols do have a big advantage: They work from behind NAT routers and firewalls.

When you use various WiFi hotspots, chances are most of the hotspots will be behind either a NAT router or a firewall, so it is kind of an important feature. Even your own home network is probably behind a so called broadband router (NAT router). At least mine is. Very few ISPs give out more than one IP to its customers, so there is really no way to get around that.

SIP is close to impossible to get to work properly behind any kind of firewall or NAT router, even if you happen to have full access to the router/firewall.

[Guest kam_]

SIP is close to impossible to get to work properly behind any kind of firewall or NAT router, even if you happen to have full access to the router/firewall.

Abosultely NOT true. SIP works VERY easily behind NAT. That's exactly what registration is for. It keeps a connection up to the regstrar and so can initiate outbound connections which the proxy can use to route the RTP packets.

This is actually a main reason why people prefer it over H323, but in response H323 now also can handle registrations.

Most unconfigurable clients will detect NAT and force proxy through the registrar (as described above, outbound connections). Configurable clients allow direct RTP to the end point. Most users who don't understand how it works get confused, and don't configure this section - then complain SIP doesnt work behind NAT. Not all SIP registrars support this mechanism, although nearly all do.

The Finarea SA group of companies (sipdiscount, voipdiscount, voipbuster etc) have some quirks with their SIP server, but do support this mechanism, and it DOES work.

EDIT:

Incidentally these people don't make their 'own' protocol for easier conectivity. They make it to force you to use their client, and only their client on their service. This way they can scroll advertisements across the client, and also charge you whatever rates they like.

Alot of the protocols i've seen are ALOT worse than SIP, and some are marginally better (because they support unicode messaging).

One other MAJOR thing to consider is that with SIP the you can negotiate codecs depending on QoS, or even user configuration. A lower rate codec could be chosen on a slower connection. The other services like SkyPE use unconfigurable codecs which don't suit many connections out in the 'wild' i.e. not on a home internet connection.

Edited April 20, 2006 by kam_

[Guest ampz]

Abosultely NOT true. SIP works VERY easily behind NAT. That's exactly what registration is for. It keeps a connection up to the regstrar and so can initiate outbound connections which the proxy can use to route the RTP packets.

And this works without having to forward any ports in the NAT router?

[Guest kam_]

And this works without having to forward any ports in the NAT router?

Yes, thats the whole point.. Typically NAT routers allow ALL out. The client registers with the registrar through port 5060 (outbound from client to port 5060). It keeps this connection (socket) up, and the registrar can respond on the same socket.

If an incoming call comes in, the registrar sends a message to the client on that same socket saying connect to me on port xxx for the RTP media (voice packets). The client connects (outbound) and they start a 2 way stream on that socket.

For an outgoing call, the client sends a message asking for a free port, and connects to that port from its side. It then uses that port for RTP.

A different mechanism proxy's everything through port 5060, thus requiring only outbound 5060 to be enabled on the firewall/router. Not all SIP proxy's support this tho.

If 5060 isn't enabled on the firewall, or outbound allow all isnt enabled on the NAT router, then you'll have to VPN to somewhere where it is - unless of course the VPN port isnt open either!

Anyway you'd have similar problems with Skype, Woize etc too if allow all out wasnt configured on the router.

[Guest ampz]

Yes, thats the whole point.. Typically NAT routers allow ALL out. The client registers with the registrar through port 5060 (outbound from client to port 5060). It keeps this connection (socket) up, and the registrar can respond on the same socket.

If an incoming call comes in, the registrar sends a message to the client on that same socket saying connect to me on port xxx for the RTP media (voice packets). The client connects (outbound) and they start a 2 way stream on that socket.

For an outgoing call, the client sends a message asking for a free port, and connects to that port from its side. It then uses that port for RTP.

A different mechanism proxy's everything through port 5060, thus requiring only outbound 5060 to be enabled on the firewall/router. Not all SIP proxy's support this tho.

If 5060 isn't enabled on the firewall, or outbound allow all isnt enabled on the NAT router, then you'll have to VPN to somewhere where it is - unless of course the VPN port isnt open either!

Anyway you'd have similar problems with Skype, Woize etc too if allow all out wasnt configured on the router.

This would not solve peer to peer SIP connections unless all the data is routed through the registrar.

Actually, Skype does not require all out to work. In a pinch, Skype will use just outgoing TCP on port 80.

They set up multiple outgoing TCP connections to different relays so that the audio stream is more or less unaffected if data on one TCP connection is temporary delayed due to packet loss.

I'am not saying that I like Skype in any way, I don't, but it is unfair to say that it has similar problems.

Anyway, I'd like to give this a try again. Is there any free and troublefree SIP registrar + SIP client combination you would recommend? Support for video would be nice.

Edited April 20, 2006 by ampz

[Guest kam_]

This would not solve peer to peer SIP connections unless all the data is routed through the registrar.

SkyPE doesn't work peer to peer either. It rely's on a central server or servers to 'proxy' the connection.

IP is IP, if you did peer to peer with skype, you would need to port map port 80 on the destination side.

So to compare apples with apples we are comparing skype proxy against sip proxy. I see no real difference so far

Actually, Skype does not require all out to work. In a pinch, Skype will use just outgoing TCP on port 80.

Well the same can be said for sip proxy. The only point i will concede here is that port 80 is nearly always open for web, wereas port 5060 may be intentionally blocked. Having said that, you can use port 80 UDP or TCP with every SIP server i've seen - granted you need to configure it

They set up multiple outgoing TCP connections to different relays so that the audio stream is more or less unaffected if data on one TCP connection is temporary delayed due to packet loss.

I'am not saying that I like Skype in any way, I don't, but it is unfair to say that it has similar problems.

Well thats just plain silly marketing nonsense! VoIP uses UDP because by the time you've discovered you've dropped a packet and got the retransmit, its too late to use the data. All this multiple TCP streams means nothing in terms of VoIP. Its posible with SkyPE to SkyPE it buffers the data and plays it back like streaming music - there it may have some value, but you would have delays in the voice.

It sounds to me like all this talk of TCP streams is just load balancing, and i'm sure your one voice stream uses 1 single socket through the least loaded SkyPE server at the time.

Besides, all the calls terminated to real physical phones will go through E1, SIP or H323/H245 where 'TCP' and retransmit means nothing.

Anyway, I'd like to give this a try again. Is there any free and troublefree SIP registrar + SIP client combination you would recommend? Support for video would be nice.

Sure download Sjphone (for PC) to test it. Use one of the Finarea companies as a test account (say voipdiscount). All you should need to do is ensure port 5060 out is allowed on your firewall.

So we are comparing a proxy to proxy. SIP proxying (like finarea servers do - point to the stun server) works exactly like you're saying

[Guest ampz]

Thanks for the suggestions but, well, I could not make it work.

SJphone registers at voipdiscount/sipdicount, and it detects that I have a "symetrical NAT".

But when I try to call my landline using sipdiscount's free trial account I get a "403 forbidden".

I also tried registering another voipdiscount account on a second computer without a NAT router. Tried calling this new user from the computer behind the NAT router. The the signal goes through, but it is never acked.

I also tried enabling DMZ in my NAT router so that SJPhone detects a "Full Cone NAT". No change.

It is getting late now. I'll have to continue tomorrow.

[Guest kam_]

Create a voipdiscount username, and use it with sip.sipdiscount.com or sip1.sipdiscount.com, port 5060 of course.

For some reason the demo accounts don't work with the rest of them.

EDIT:

SJlabs have released a beta WM5 version here:

http://www.sjlabs.com/preview/ce/

again for PPC, but it apears to run on my SP5. Just checking now if its usable at all

Edited April 20, 2006 by kam_

[Guest kam_]

Well it works.. i just helped flodis79 configure it, and make a sucessful test call.

With a reg key you should be able to load in the skin too. I'll leave someone else to play with it further.

[Guest mod25]

@kam_

could you provide files (maybe zip) which i can extract to tornado also registry because i just have the tornado and not an wm5 ppc.

Or could you poste an detailted howto do it?

mod25

[Guest kam_]

1) go to sjlabs.com and download the PC version

2) Edit the SIP PC to PC profile and put in your account details for your sip provider.

3) Make a test call - YES from your PC, to be 100% sure you settings are correct

4) Extract the files in the zip to your phone. You should have the exe, dll and skin file in any directory you want to put it in, and the profiles (.ini files) in sub directory Profiles. For example:

\Storage Card\SjPhone\Default.skn

\Storage Card\SjPhone\SJphone.exe

\Storage Card\SjPhone\SJphoneSetup.dll

\Storage Card\SjPhone\Profiles\H.323 direct.ini

\Storage Card\SjPhone\Profiles\SIP direct.ini

NOTE to mods: The cab file is freely available at http://www.sjlabs.com/preview/ce/ (a link from their download page). I've extracted the files from it using WinCE manager and put them here. If thats not ok, then just delete the atachment and everyone can just do that step themselves!

5) copy the .ini file from your PC (c:\program files\sjlabs\sjphone\profiles i think) to the profiles directory on your phone.

6) launch sjphone on your phone

7) Hit left soft key and scroll to services. Select PC to PC (sip).

You may have to hit the left soft key and shutdown, then restart sjphone

Once its started, you can scroll to the dial box, and dial the number (hold * to change to numbers first), then scroll down to dial. Unfortunately you can't delete a digit you enter so its tricky!

Here's an export of the reg keys from the cab file:

REGEDIT4

[HKEY_LOCAL_MACHINE\Comm\Afd]

"DgramBuffer"=dword:00000008

[HKEY_LOCAL_MACHINE\Software\Microsoft\Shell\URLProtocols\callto]

@=""

[HKEY_CLASSES_ROOT\callto\DefaultIcon]

@=hex(7):25,49,6e,73,74,61,6c,6c,44,69,72,25,5c,53,4a,70,68,6f,6e,65,2e,65,78,\

65,2c,31,00,00

[HKEY_CLASSES_ROOT\callto]

"URL Protocol"=""

[HKEY_CLASSES_ROOT\callto\Shell\open\command]

@="%InstallDir%\\SJphone.exe /callto %1"

[HKEY_LOCAL_MACHINE\Software\SJLabs\SJvoip Project\SJphone]

"App"="%InstallDir%\\SJphone.exe"

[HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone]

"key1"="GQVV0-3OUI3-APJB4-TKEJ1-A1EGG"

"key2"="GQVV0-3OUI3-APJB4-TKEJ1-A1EGG"

[HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\General]

"AddressBook"="\\%CE5%\\SJphone.xml"

[HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\Call Logs]

"CallLogFileName"="\\%CE5%\\SJphone.clm"

[HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\Skin]

"SkinDllFileName"="%InstallDir%\\Default.skn"

"UseSkin"=dword:00000001

[HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\Profiles]

"ProfilesDir"="%InstallDir%\\Profiles"

"PersonalDataDir"="%InstallDir%\\Profiles"

"ActiveProfile"="C0DFB9E1-79A0-4749-844F-18EC3DE3629D"

[HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\RTP]

"RTPService"=dword:00000032

You don't need ANY of them to get it to work, but you may want to try to get the skin working by adding this one:

[HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\Skin]

"SkinDllFileName"="%InstallDir%\\Default.skn"

"UseSkin"=dword:00000001

Of course replace %InstallDir% with where you install sjphone

Oh and here's a list of the identifiers in case you're wondering what they mean:

CE1 \Program Files

CE2 \Windows

CE4 \Windows Startup

CE5 \My Documents

CE11 \Windows\Start Menu\Programs

CE12 \Windows\Start Menu\Programs\Accessories

CE14 \Windows\Start Menu\Programs\Games

CE15 \Windows\Fonts

CE17 \Windows\Start Menu

SJPhone.zip

Edited April 21, 2006 by kam_

[Guest mod25]

@kam_

no chance to edit SIP Profile PC to PC because i can't add my SIP Proxy Server...

Maybe i do something wrong? Please help.

mod25

[Guest kam_]

@kam_

no chance to edit SIP Profile PC to PC because i can't add my SIP Proxy Server...

Maybe i do something wrong? Please help.

mod25

Follow this, its for sipx, but it should be the same regardless of your provider:

http://sipx-wiki.calivia.com/index.php/How...phone_with_sipX

Instead of creating a new profile, edit the existing PC to PC one.

You can create a new one if you want, but i'm not sure sjphone will pick it up unless you add the reg key to point to the profiles - but try it and let us know!

Edited April 21, 2006 by kam_

[Guest kam_]

Ok so you can activate the GSM codec by fixing the codec keys, and disabling G711a and G711u.

If you run sjphone once, then exit, most of the directories get created and you only have to put a few key values in.

Also fixing the Skin reg key will load the skin (although if you enable the included default skin, you can't scroll to the hangup button! - but since you can't delete the number anyway, you'd have to exit and reload sjphone to make a new call). I guess somone could make a much more intuitive 'smartphone' skin if they have time.

Here's some screenshots of how the registry keys should look. You can download a free PC based registry editor from here (Alot easier than trying to create keys on the device itself):

http://mobile-registry-editor.en.softonic.com/ie/47279

Anyway, by using the GSM codec, you can get this to work over GPRS with acceptable quality - unless the cell gets really busy.

P.S. ignore the wierd keys begining 333 etc.. its a quirk with my old shitty ipaq 1910 running PPC2002 (which i used to check how the keys should really look with a proper Sjphone install)

Edited April 24, 2006 by kam_

[Guest ampz]

Tried it.

It works, but as previously mentioned, some functions like "hang up" and "backspace" cannot be accessed due to the lack of a pointer device.

The program is also rather slow.

The audio quality and lag when using GPRS is too poor to be useful. Have not tried WLAN yet.

[Guest uknutz]

Its working on Qtek 8300 using the instructions above. Connected to voipstunt sip server over wifi with no problems at all. Created a new profile on the PC then copied it into the correct directory on the qtek and it found it straight away. Not tried GPRS yet (dont think i will bother) and can only get sound through speakphone ;)

If only the backspace key worked :P

Jim.

[Guest illumin8]

Hi all,

I finally managed to run SIP client "x-pro" pocket PC edition on i-mate SP5 (qtek 8310) and i-mate SP3i (qtek 8020).

1 - Get the "x-pro.cab" file. Install it on a PPC then copy two files "x-pro.exe" and "AES_PPC_DLL.dll" to SP5 /windows directory or /storage/windows for sp3i;

2 - Set up your SIP accounts on the PPC running "x-pro". Activate x-tunnels if you intend to use non-wifi connexions. Then, using Resco Reg Editor, export "/HKEY_CURRENT_USER/Software/xten.../X-PRO" to x-pro.reg file, and copy this file as well to your SP5 or SP3i;

3- Create a shortcut to "x-pro.exe" in /windows/Start Menu/ (Storage/windows/Start Menu for Sp3i);

4 - Run Resco Reg Editor on your SP5/3i, import "x-pro.reg" file, reboot and get ready.

5 - connect SP5 to wifi (connect Sp3i to internet via bluetooth and activesyc 4.1)

6 - launch X-PRO from the "Start" menu and wait untill it is connected to you SIP gateway.

7 - simply dial the numbers from the smartphone dialpad, connect with a press on the joystick.

8 - You're done !!! Quality of sound is acceptable on SP5, very PTT like from SP3i.

9 - No need to overclock or whatever, it doesn't seem to have any effect on quality.

10 - Log window can be accessed by a long press on "#"

11 - Any comments, please feel free to post.

Screenshots attached.

Hi,

Thanks for this. Will buy and try the X-Ten PocketPC client ... BTW, do you know of an open (as opposed to closed clients from Barablu & Woize) and free SIP softphone that runs on WM5 smartphones like the QTEK 8300/8310 i-mate SP5/5M?

[Guest flodis79]

Hi,

Thanks for this. Will buy and try the X-Ten PocketPC client ... BTW, do you know of an open (as opposed to closed clients from Barablu & Woize) and free SIP softphone that runs on WM5 smartphones like the QTEK 8300/8310 i-mate SP5/5M?

Don't buy X-ten - i havent been able to use it on Qtek... Use instead the sjphone ppc version.. It works fine with voipbuster!

[Guest abv]

Don't buy X-ten - i havent been able to use it on Qtek... Use instead the sjphone ppc version.. It works fine with voipbuster!

Hallo Buds, I was away for two weeks and I am really pleased to discover SJPhone issue.

I have no problems using X-PRO for PPC (

[Guest flodis79]

I have no problems using X-PRO for PPC (

[Guest abv]

What sip provider do you use with x-pro? I havent been able to connect to any provider with the xten ppc.

Freeworlddialup.com