Getting VoIP Working On E90 and Other Nokia S60 Phones
Configuring VoIP on Nokia phones can be a real nightmare. Moreover, some VoIP services work just fine, with some you may get a partly working solution (i.e. one-way audio), yet with others you can't even register with the service. Registration Failed is a painfully known message for Nokia phone users.
Occasionally, on various forums, you can see messages posted by disappointed users and some posts are really rude. But I can understand these people. When you pay a grand for a phone and you can't even register with a service provider, that is not good. For some people, support of VoIP becomes a key feature when making decision which phone to buy. For example, when I first got Nokia E61, I must admit, I bought it because I hoped to use it as a VoIP phone. Nokia E90 Communicator is a great device and it is very pity when you are not able to use it with your VoIP services provider.
But I have good news! This post will help you to get ANY VoIP service working on your Nokia S60 phone. No kidding. Just read carefully and then follow the instructions and let the magic happen.
I mainly use two VoIP services: Stanaphone and SipGate. I was able to get SipGate working both on E61 and on E90 right away without any problems. However, no matter what I did, I could not get Stanaphone working. I created accounts with almost every VoIP provider on the Internet from GizmoProject to Voxalot to PBXES.org to test and troubleshoot this problem.
If you configure SIP Settings in your phone according to the provider's details, most of the services simply DON'T WORK! Frustrated users even say that the VoIP feature in Nokia phones is a beta product. However, that is not true. After you get your VoIP service working on your Nokia, it will work smoothly and your Nokia phone will turn into a truly Internet Phone.
What Causes Registration Failures?
Problem Number One - Incorrect REALM Value
When configuring SIP settings, specifying correct REALM is very important. If the REALM used by the service provider is different from what you specify in SIP Settings, registration will fail for sure. With Windows based softphones this is not a problem and because of this, the service providers sometimes may not even publish the REALM or publish incorrectly on their sites. For example, look at the settings that Stanaphone provides on the screen shot below.
As you can see, according to their site, you must specify sip.stanaphone.com as a Domain/Realm or Proxy Server value. Unfortunately, if you specify sip.stanaphone.com in the Realm field in your S60 phone, it won't work. Correct realm is sip.stanaphone.com.
I have tried stanaphone.com and many other values as well. For example, sometimes just "asterisk" works. Why? Because it is a default REALM value in Asterisk - a popular VOIP software that many VoIP providers use. But nothing worked.
The problem is that there are quite many SIP settings and there are so many possible combinations of different settings (like specifying TCP instead of UDP, switching the Security setting on or off, etc) that you cannot get it working by chance.
So, how did I manage to find out a correct realm for Stanaphone? It was not too difficult. I have been able to successfully use the service with X-Lite. So, I switched logging on and then looked up the log file. As you can see on the screen shot below (it's a part of the log), the correct realm is just stanaphone.com.
Problem Number Two - STUN
The problem with VoIP in Nokia phones is associated with how the phone used STUN. As you are well aware, you cannot specify a STUN server in the SIP Settings and that is the problem.
As far as I know, earlier version of the VoIP part in Nokia did not use STUN at all and because of that you could not use the VoIP-enabled Nokia phones with many providers. However, in the latest models STUN is incorporated, but the problem is how the phones uses the STUN server info.
One way that is used in softphones is to make a DNS lookup and that's how Windows softphones do and that is in my opinion how Nokia phones work. As you can see from another part of the log, the default STUN server setting did not work. Now this makes clear why Nokia phones work with some VoIP providers and don't work with others. If the STUN server coincides with the one that Nokia phones try to use, the service works.
Logically, a solution to this problem would be to be able to specify correct setting for STUN. However, as I have already mentioned you cannot configure STUN setting from the SIP Settings section. But, there is a tiny but extremely useful application from Nokia that allows you to change some very internal SIP settings. That is the application that does the magic. So, download it quickly from here http://sw.nokia.com/id/d2d27e6c-bd52-4534-9aa6-19e606b80709/SIP_VoIP_Settings_v1_0_en.zip and install on your phone.
In order to change the STUN settings, you must first create the SIP Settings. Stanaphone settings are shown below:
After you have created the SIP Settings, launch the application. The first screen presents you with 2 choices: VoIP Services and NAT Firewall Settings. Choose NAT Firewall Settings.
Then you will see two choices again: Domain Parameters and IAP parameters. Choose the first.
You should see your VoIP profile there. Choose it.
Finally, you can specify the STUN server there. If you don't know STUN server for your VoIP provider, you can specify this: stunserver.org. This is a publicly available free STUN server. You must also specify the default 3478 port.
On the final screen shot you can see that the Stanaphone profile got registered successfully.
"You should see your VoIP profile there."
That's where it goes wrong for me...
You see no profile there? I didn't see either but try this. In the public name put fully qualified name. For example, if your Stanaphone number is XXXX, put XXXX@sip.stanphone.com. For SipGate you must put in a XXXX@sipgate.co.uk, etc. After you put the public name in that format, you will see it in the NAT settings for sure! Actually, that is how SIP presents each user.
THANK YOU!!! The key for me was using X-Lite to grab the realm... I'd even talked to the tech support guy at my VOIP provider, who (obviously) had given me the wrong info.
FANTASTIC!!!
I've got this working but the E90 seems to still randomly fail to register properly.
Thanks a lot, you just save me a lot of time!!!
The sip voip settings application doesn't work on e70-2's, or atleast mine. It installs without problems, but won't run afterwords, no errors, just back to the program selection list.
Also, mention was made to use stanaphone. Where would I download stanaphone for a Nokia E90 communicator?
Hi Vman,
The beauty of E90 (as well as VoIP enabled S60 phones, mostly E series) is that you don't need to download any software.
E90 already has a built in support for StanaPhone and many other VoIP (SIP based) providers.
All you need is to configure StanaPhone on your E90 and you're done.
The problem was that it used to be quite complicated to setup VoIP on E90 and other Symbian phones. But with the configuration tool this task has become easier, though not as easy as many people would wish.
Hi again. Thanks for the previous reply george. I have tried the settings in the E90, but when I get to the section "domain parameters" in "NAT Firwall settings" it says no settings. I cannot create a parameter either as it says "There are no domains defined. define public username with domain in SIP settings" I think I have entered all the settings but have obviously missed on something. Any pointers?
Hi George
Still trying to get this E90 to work. The VOIP service I am using is Kiwilink (I live in NZ). I have the settings for it but not sure how they fit in with the stanaphone settings you have suggested. I could send you the email I received from themwith the server details etc.
I have already posted this in comments :) Here it is how to solve it
In the public name put fully qualified name. For example, if your Stanaphone number is XXXX, put XXXX@sip.stanphone.com. For SipGate you must put in a XXXX@sipgate.co.uk, etc. After you put the public name in that format, you will see it in the NAT settings for sure! Actually, that is how SIP presents each user.
Hello Vman,
Yes sure, please send the details and I will try to provide you with either screen shots or some other help to get it working. Please send to: george_sazo (at) yahoo (dot) com.
I will try to respond as soon as possible.
That's great.
It works also with Panasonic KX-TDE PBX series. The REALM value expected by Panasonic is 'Registered Users'.
The Eagle.
Hello,
I managed to follow the instructions and got StanaPhone to register through my Nokia N82 (many thanks) but when I try to make an Internet call I get "Connection Error". What could be the reason.
The strange thing is that I can receive calls to my Nokia N82 through StanaPhone but I cannot I get the above error when trying to do Internet Calls.
Please help.
I set up really quickly after abandoning freecall in favour of sipgate.
Thanks for your work, excellent.
Peter, you're welcome. I'm glad if anything in this post has helped you.
I've been using SipGate for quite long and I love their service. It's very reliable and affordable.
George
I´m also having the connection error, when I´m trying to dial, anyone know how to fix it?
what number exactly do you mean by Stanaphone number XXXX i think thats where the problem is coming from cause i set up everything just as you suggested the stuf still keep saying unregistered after the setup on my E90
I've made a screen shot of a Stana's page. XXX in a red circle is where you find that number.
Thus, XXX is NOT THE SAME as the phone number.
You can get the screen shot here http://i216.photobucket.com/albums/cc74/georgesazo/SS/stana_screenshot.gif
Hope this helps.
I am trying very hard to configure stanaphone on my N82, but still not working. Can anyone please help.
I am trying very hard to configure Stanaphone on my S60 Nokia N82. Can someone please help. Thanks. Naveen
Hi! I can use my E-51 with different voip reg-s (Voipcheeap,etc)well, but after 5-10 minutes of registration people can't call me back.I can use stun server with the port 3478, I tried to give various values to the TCP NAT bind refresh, UDP NAT bind refresh, and CRLF refresh "on" or "off", but after a few minuts nobody could call me. Please, help me, what's the correct setting there? Thank you.
dear i use freecall and i have tried every thing for it but cannot register the service it works fine with fring but the quality is not good so please help me for the same
SIP on 3g or GPRS. Hi, I have installed successfully my SIP providers details on my Nokia E90, it works very well when connected to a local wireless LAN, however I cannot configure it too work with GPRS or 3G [I live in the UK], I have a permanent connection to the internet via GPRS or 3G the problem is it will not by default try these connections rather than the local wireless lAN. It is not in the interest of the mobile phone service providers to open up SIP to mobile phones. Any ideas Pls
DGPS all I had to do was change the default access point in the SIP Settings from WLAN to my GPRS connection and it registered fine.
BEAUTIFULL EXPLANATION!!!
Thanks a lot for your post. certainly the best on the web (and I visit many...)
But, just a single observation: In my case, the X-Lite log appears two "Realm=" values. One, in the beggining of te log (realm=sip.wpvoip.com.br, BUT other more ahead (realm=asterisk). Only the second works.
I think that is can be a mistake on the asterisk configuration.
Happy 2009!
Oh... Another comment:
To many Nokia firmware, the correct application that 'does the magic' is another if the SIP cliente is 2.x or 3.0.
In this page on the Nokia´s site are the others aplications for these versions:
http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settings.html
And on this page (Nokia´s site too) has a tabl with the various voip client versions:
http://www.forum.nokia.com/Resources_and_Information/Explore/Mobile_Technologies/VoIP/Nokia_VoIP_Framework/VoIP_support_in_Nokia_devices.xhtml
Sorry about my english.
Any doubts -> romano@copiadorasatelite.com.br
Well.... Me again... :-)
Another question is: If your VoIP service do not recognize any of the anabeled codecs, the login will be falty.
So, make available all of the six codecs available by the Nokia SIP Configuration Tool.
Thanks again.
Renato Romano - São Bernardo do Campo/SP - Brazil
Great post! You saved me soooo much work!
finally I have voip on my phone. It was the releam. I used tcpdump to investigate the sip conversation.
Thank you!
We have successfully tested our PQSCall Mobile for Symbian dialer from blocked regions and it worked just perfect. More information about the PQSCall Mobile can be found here
http://www.pqscall.com/PQSCall_TCT_Signed.sis
The softphone supports following codecs: g729, g711
The softphone supports VoIP tunnel technology and can work from blocked areas or from behind firewalls for both incoming and outgoing calls.
The softphone can connect to Internet using Wifi, GPRS, EDGE or UMTS - user is allowed to select which method when starting the application
Vippie! mobile can work with voipswitch and also with 3rd party sip servers.
The softphone is linked to an IP address of a PQSCall (or a sip server) or to DNS name for example sip.pqscall.com
Certificate:
Each softphone is digitally signed and certified by Symbian/Nokia consortium. The certificate is required per each copy, for example if a provider changes IP address then the PQSCall installation file will have to be certified again. Therefore recommended is to use DNS names which remain the same even when the IP of server is changed.
Obtaining the certificate is paid. For first time it is included in the price of the dialer. For every next certificate it is to be incurred by a provider (our customer).
Certification process takes about 5 days (sometimes less) and is carried out by Symbian OS.
License:
License is per one IP address or DNS name.
There is no limits in number of endusers nor number of conncurent calls.
The softphone can be placed on provider’s server so all endusers can download it freely directly from website.
Installation:
The installation package comes in one file PQSCall_TCT_Signed.sis
The file can be downloaded directly on the phone and then installed or it can be download on a PC and uploaded to the phone thru for example USB port.
Backup:
For backup - failover if the main server does not work, recommended is to use DNS. PQSCall Mobile Dialer is pointed to DNS name for example sip.PQSCall.com and then on the DNS server the provider has to set backup IP address along with the main IP address under one name but with different IP. When the main server is down the DNS will direct incoming connections from the dialer to the backup server.
Also Network Load Balancing can be used for this purpose. For more details please contact us
Features:
connectivity thru Wifi, GPRS, EDGE, UMTS - the softphone asks user to select the prefered method from the list of available (defined on the phone) internet access points.
The window with the list of access methods is shown during the softphone start
making and receiving SIP calls
making and receiving SIP calls from behind any VoIP blockades/firewalls.
It uses our propriatary solution called VoIP tunnel which is also used in our PQSCall windows edition and previous softphone SIPlink and has proved very efficient throughout last 4 years since it was released. VoIP tunnel works without any problems from countries like UAE and all others where voip is blocked.
native address book from mobile phone fully integrated with the PQSCall mobile. User can select a contact from the address book and make call
redial button
real time call status messages
All above features can work with any SIP server (not only PQSCall), for tunnel to work it is required to have a voip tunnel server to be installed on the sip server side. Please contact sales@PQSCall.com for more details. In PQSCall server the voip tunnel is already included as a part of the main package
SMS sending (only for PQSCall server), available in April 2009
Jay Patel
Senior Sales Director
Email : Sales@PQSCall.com
Direct Tel : +1.302.2615201
URL : http://www.PQSCall.com
IM Yahoo, MSN, GoogleTalk : PQSCall
You absolutely ROCK!!! Thanks so much. This was driving me INSANE!!!
I did "asterisk" for the realm, and everything is now A-ok.
Cheers.
Dulwithe
AAAAA!!!!! Nice work
You can also use zoiper on the pc. (Zoiper is working on mobile phones as well, but not for symbian yet).
Get it from http://www.zoiper.com
Good information. Very understanding!