Nokia Cell Phones with VoIP/SIP Support
Nokia S60 Phones with a SIP stack
- Nokia E51
- Nokia E60
- Nokia E61
- Nokia E61i
- Nokia E65
- Nokia E70
- Nokia E90
- Nokia N80ie
- Nokia N81 (8GB)
- Nokia N82
- Nokia N95
Nokia Series40 Phones with a SIP stack
- Nokia 6300i
Notes about Nokia's VoIP support
- Nokia phones have varying revisions of their VoIP stack on their phones.
- Nokia maintains a chart of VoIP support across different Nokia devies
- S60 VoIP Release 1.0 does not support STUN or NAT traversal. This applies to the E60 and E70.
- For best performance, be sure to upgrade the phone's firmware to the latest revision. Updated versions of the VoIP Releases fix many issues.
Configuration
Asterisk Configuration
sip.conf
[nokiaphone]
username=<username>
type=friend
secret=<password>
host=dynamic
context=<outgoing phone context>
qualify=60000
Qualify timing. While the phone is in power save mode (screen saver), it responds more slowly. It appears that during power save mode the response time can go up to about 30 seconds. By default qualify in asterisk is sent every 60 seconds (chan_sip.c) and expects a response within 250 (or timed specified by the qualify parameter). Because of this, asterisk will see the phone as unavailable as soon as power save mode comes on. In order to keep alive the Nokia phones and have inbound ringing always working, set qualify to at least 30000 (30 seconds). By doing this, the nokia phones should be reachable behind most firewalls. This was tested using a trendnet router and a linksys router. On the linksys, the phone was reachable after 9 hours of power saving and on the trendnet for 4 hours.
Basic Phone Configuration
The latest official VoIP configuration manual from Nokia
Phone configuration consists of three steps:
- Define an Access point
- Tools -> Settings -> Connection -> Access points
- Select Options with left softkey and choose New access point
- Fill in appropriate parameters to connect to wireless network
- Create SIP Profile
- Tools -> Settings -> Connection -> SIP settings
- Select Options with left softkey and choose New SIP profile and Use default profile
- Populate as shown in tables below
- Proxy server can remain empty
- Create an Internet Telephone Profile
- Tools -> Settings -> Connection -> Internet tel.
- Select Options with left softkey and choose New profile
- Set any name and select newly created SIP profile
| Parameter | Value |
|---|---|
| Profile name | <profile name> (anything) |
| Service profile | IETF |
| Default access point | <access point defined above> |
| Public user name | sip:<username defined in sip.conf>@<hostname/IP of asterisk server> |
| Use compression | No |
| Registration | Always on (necessary to receive incoming calls, otherwise can be set to when needed) |
| Use Security | No |
| Proxy Server | no need to configure |
| Registrar server | (see below for configuration) |
| Parameter | Value |
|---|---|
| Registrar server address | sip:<hostname/IP of asterisk server> |
| Realm | asterisk (or change to match if overwritten in sip.conf) |
| User name | <username defined in sip.conf> |
| Password | <password defined in sip.conf> |
| Transport type | Auto or UDP |
| Port | 5060 |
Advanced Phone Settings
Starting with VoIP Release 2.0, Nokia released a tool called SIP VoIP Settings Tool which allows advanced tweaking of many VoIP parameters.
Of particular importance is the ability to set a STUN server and define NAT firewall traversal parameters.
The STUN server is set on a per asterisk server basis and the NAT traversal parameters are set on a per access point basis.
SIP VoIP Settings Tool and its manual can be downloaded from Nokia
Setting STUN server
- Run SIP VoIP Settings Tool
- Installations -> SIP VoIP Settings Tool
- NAT FW settings -> Domain parameters -> <asterisk server>
- Set STUN server name to hostname of STUN server and set STUN server port to 3478 (unless not running on default STUN port)
Setting NAT firewall traversal parameters
- Run SIP VoIP Settings Tool
- Installations -> SIP VoIP Settings Tool
- NAT FW settings -> IAP parameters -> <target access point>
- TCP Nat bind. refresh and UDP NAT bind. refresh are in seconds. STUN retransmission is in milliseconds.
Implementation Issues
Traversing Multiple Distinct WLANs
Nokia's current implementation of SIP profiles binds an account to a given WLAN access point.
If the phone is used across multiple WLAN networks, then there should be one SIP profile entry for each unique ESSID that it connects to.
When defining the Internet telephone settings, all SIP profiles should be bound to a single Internet telephone profile.
Traversing the same WLAN across multiple subnets
As is the case with many portable devices, the Nokia phones suffer from connectivity problems when traversing a single WLAN (same ESSID) that is broken up into different subnets because of IP limitations or routing simplicity.
The typical use scenario is in a large university where different regions of the campus have different subnets, but all access points broadcast the same ESSID.
This is arguably a network design issue instead of a client connectivity issue. Nonetheless, the issue is quite prevalent and can be quite debilitating.
The Nokia phones, particularly the ones that are pre-S60r3 FP1 (see device specification matrix), have particular difficulty releasing/renewing their DHCP leases when moved to an access point on a different subnet.
MWI
Despite Nokia's claim to the contrary (see feature matirx), no VoIP Release versions support message waiting indicators. This prevents notification of new voicemail on the asterisk server.
This has been a long-standing bug, but Nokia has yet to fix it.
There is a potential fix for this mentioned here by sending the urgent message count in addition to the messages waiting in the same notify packet, but this has not been confirmed.
SIP Messaging (via MESSAGE method)
There have been mixed reports of messaging support with Nokia devices.
Functionality appears to completely vary with phone, working well on some devices and not at all on others.
There has been reported success with E61i and E70, but it does not appear to work with E51.
External Resources
- Mobile VoIP Forum
- How-To configure Nokia S60 phones for VoIP / SIP
- [http://forum.voxilla.com/nokia-voip-support-forum/
- Setup Asterisk Trixbox with nokia E60
- Using the Nokia E-series Phones with Asterisk
- Configure VoIP-Capable Nokia Phones to Work with Localphone
Where to Buy
- Amazon.com
- Buy.com
- Voxilla Store - Nokia Phones - Authorized Nokia Reseller
- Wildix in France France
- Wildix in Italy Italy
- Wildix in Spain Spain
- Wildix in Ukraine Ukraine
Page Changes
Nokia E51 Freezes
Re: Strange Problems with Netfilter Modules
I am having the same ploblem. Is wierd, i register in asterisk using my notebook(wifi) but the E65 runs into timeout. Did you solve the issue?
Nokia E51 Freezes with Asterisk when called
The phone rings, but freezes when I answer the call. The only thing that helps is removing the battery.
I run version 100.4.20 RM-244
However, I have it working all well via sip.xs4all.nl. Both incoming and outgoing calls :-/
Nokia E51
But I could not connect to my SIP server or I couldnt use voip Phone yet.
My Wlan is working fine and connecting all servers.
I am expecting a detailed reply in this thread..
Rgds
N95 - plus sign in international calls
However, many of my phone numbers in the address book starts with a plus sign followed with the<br>
international code for my country and then the telephone number. <br>
Asterisk doesn't seem to like this. <br><br>
Anyone found a workaround?
N95 - settings?
However, I have now fetched the full settings tool from Nokia and am wondering if there is any way I can set it up to work over WCDMA and possibly on other wireless networks if a STUN server will get round it - I am aware that will not always do it but it may help.
I am not sure why, but nothing happens at the Asterisk end when I try over WCDMA.
N95 - settings?
However, I have now fetched the full settings tool from Nokia and am wondering if there is any way I can set it up to work over WCDMA and possibly on other wireless networks if a STUN server will get round it - I am aware that will not always do it but it may help.
I am not sure why, but nothing happens at the Asterisk end when I try over WCDMA.
Nokia E51 (even through NAT)
I tested it over LAN and WAN connections and I can report that E51 works well even through NAT.
Btw, if you have a dynamic WAN IP address (like I have) make sure you add an "externhost" directive in your sip.conf. After some playing around I got this solution:
In sip.conf I added the following:
\general\
externhost=yourhost.dyndns.org
externrefresh=120
localnet=192.168.1.0/255.255.255.0
"externhost" and "externrefresh" work on Asterisk 1.2.x. "externrefresh" tells asterisk how often (in seconds) it will resolve the "externhost". "localnet" is used to define your LAN IP addresses (and/or other addresses that do not need NAT).
The only annoyance I found on E51 is that I have to "force connect" wireless before I try the SIP connection. I have to "Start web browsing" on a wireless connection, open some web page (to force the device to connect to wireless), then exit using the home button (without closing the browser) if I want to make a SIP connection. Otherwise my conenction to Asterisk always fails. I guess it has to have a wireless connetion already established prior to conencting via SIP. I don't know if this is by design or not, but I found no other way to get around it. If anyone knows how to establish a wireless connection on E51 without having to open the broswer, please tell.
VoIP Security Solutions
The core solution for VoIP Security and VoIP anti-blocking is VGCP (VoiceGuard Control Protocol).
It can work with any 3rd-party Softphone / ATA / Gateway / IP Phone / IADs and SIP proxy or server.
It can work in the way similar to that of SOHO router, but it only encrypts and decrypts SIP and RTP packets on link layer, not to handup these packets to IP stack for forwarding while bypassing other data packets originating from SIP terminals. In this scenario, peak throughput and minimal CPU overhead can be easily achieved.
VoiceGuard can real-time incorporate light-weight traffic for puzzling and bypassing VoIP blocking system without consuming more bandwidth and compromising voice quality. Even in some circumstance, VoiceGuard can simulate traffic behavior of universal data networking protocol such as OICQ, MSN and so on.
For more information, please refer to: http://www.speed-voip.com/index-36.html
Andy
xd.wong@speed-voip.com
andywong-01@hotmail.com
N95 and asterisk problems
Got SIP response 400 "Bad Request" back from 192.168.5.10
Where 192.168.5.10 is my N95.
If a call from another extension to my N95 this ring normally but if I try to call from my N95 I get that message.
Any one have any idea ?