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Felipe Andrade
Read more about Felipe on the Champions website.
Ovi Store Publisher Reporting
July 08, 2009
9 a.m. New York, 2 p.m. London, 4 p.m. Helsinki
Ovi Store Publisher Reporting
July 08, 2009
11 a.m. San Francisco, 2 p.m. New York
Nokia World 09
September 02, 2009
Stuttgart, Germany
Adobe MAX 2009
October 04, 2009
Los Angeles
International CTIA WIRELESS I.T. & Entertainment 2009
October 06, 2009
San Diego, California
When inquiring about a problem related to Nokia N80 Internet Edition or VoIP, please describe the problem in detail. Include all the relevant information that you have available of the problem situation, details of the system (for example, what equipment and software build is in use), and any other relevant information (for example, log writings and message monitoring). Also, describe the corrective measures that you have carried out after the problem occurred. All data may be valuable.
Settings need to be modified to match the voice-over-IP (VoIP) service provider’s network.
Service providers are strongly advised to deliver the settings either over the air using the Open Mobile Alliance (OMA) Device Management (DM) facility supported by the Nokia N80 Internet Edition or by using a settings wizard implemented as a plug-in module for the VoIP log-in application.
Please refer to the document Nokia S60 VoIP Implementation Configuration Tutorial.
The Nokia N80 Internet Edition supports OMA device management and automatic provisioning. Please refer to the document Nokia S60 VoIP Implementation Configuration Tutorial.
The only way to know interoperability is to test the whole system end to end. Even when the Nokia N80 Internet Edition is known to be fully interoperable with a certain SIP proxy, one single configuration change or WLAN access router or a firewall on the route can break it. On the other hand, any nonlisted equipment may work.
Nokia will publish a list of vendors it is conducting tests with, and any specific product interoperability can be checked from the voice-over-IP (VoIP) support.
Network providers will publish their settings or preferably provide the over-the-air (OTA) settings automatically.
All settings (SIP, VoIP) can and should be provisioned to increase usability and avoid human errors.
The SIP setting or settings are selected for a certain voice-over-IP (VoIP) profile inside the VoIP settings.
To be reachable via voice over IP (VoIP) whenever under the WLAN coverage, the "Always on" setting is preferred. On the other hand, manual registration is more convenient when you need an occasional or a temporary VoIP connection.
If automatic registration is used, the SIP client is also changed to automatical registration.
Yes, assuming the service provider uses the existing normal phone numbers as the SIP user name. The voice-over-IP (VoIP) client will add the default SIP domain automatically.
For frequently used VoIP contacts, and if they are on another SIP domain (that is, under another VoIP service provider), the SIP address should be added to the phone book. This can be done the same way as with normal cellular calls on the received-calls log.
Yes.
Yes.
Yes, assuming there is an IP route to both of them from the same WLAN access point.
Unfortunately not.
Yes.
Unfortunately not in the current software release. The registrar can shorten the expiration time, if needed. A WLAN client can be detected based on the User-Agent header.
Unfortunately not in the current software release.
Unfortunately not in the current software release.
Yes, if you have a registered SIP account. The over-the-air (OTA) and plug-in provisioning will set the Internet call as the default, and the terminal changes automatically to Internet calls if no subscriber identity module (SIM) card is inserted.
When the terminal is used without a subscriber identity module (SIM) card, the terminal will always use an offline profile. In the offline profile, the ring tone is "none" by default.
It can be changed manually via Menu -> Tools -> Profiles -> off-line -> edit -> Ringing Tone. In offline profile the ring tone can be changed only manually, not via provisioning.
Unfortunately not in the current software release.
There can be * characters in the dial string, so a call via certain SIP broker services works, but unfortunately # character is not allowed.
Certain dial string combinations starting and ending with * and/or ending with * or # are reserved for the GSM call control and thus they can not be used with VoIP (for example, *21#).
The codec order can be provisioned. You cannot edit the order yourself.
The STUN server address, either an FQDN or a direct numeric IP address, can be defined with provisioning. If not provisioned, the VoIP client will try to resolve the address with a DNS SRV query. The query is expanded from the SIP domain, that is, the hostname part of the user’s SIP URI as specified in RFC 2782 and draft-ietf-behave-rfc3489bis-04.
For example, if the configured SIP URI is sip:john.doe@my.voip.provider.com, the DNS SRV query will be performed to _stun._udp.my.voip.provider.com
No. If a call is established using a SIP account, it can be stored on the call list with the domain, if so required. Remove the domain suffix in establishing a GSM call.
A selection has been made between numbers and characters as the default method. You can change to number mode by using a long press function. If the number is in the phone book, calling is easier.
The Nokia N80 Internet Edition moves to null state immediately. All Nokia mobile devices work in a similar way.
Phone numbers in Nokia N80 Internet Edition can be entered with or without the country code. Nokia N80 Internet Edition takes the number from contacts and adds the set domain , ie. sip:0400123456@some.domain will be used.
The form may make a difference in SIP proxy, this depends on the service provider. If there is no number analysis for the SIP URI, the same number with and without country code is considered different. For example: sip:0400123456@some.domain and sip:+358400123456@some.domain may be different in some service operator's network.
If the user initiates a call by keying in a TEL URI in Contacts, the call is established using the TEL URI.
If a TEL URI is selected in the browser, the phone will ask for a voice, video, or Internet call. If the Internet call is selected, the call is initiated with a SIP URI using a number in the TEL URI.
An incoming call with a TEL URI is shown in the UI and Logs. Using a TEL URI in the Logs will initiate a call using a TEL URI.
In dual mode handsets with Nokia VoIP framework capability, an emergency call is always tried first as a cellular (GSM/WCDMA, that is, circuit switched) call if cellular coverage is available. The cellular emergency call recognition and procedure are done as specified for cellular emergency calls.
Only if cellular coverage is not available for a circuit-switched emergency call, the dual mode terminal will try to make the emergency call as a WLAN VoIP call.
The existing GSM technology is mature for such a critical service as the emergency call. The mobile VoIP emergency calling has still some unsolved technical issues and has not been widely deployed. There are also technical differences in the support for mobile emergency calls between the VoIP service providers.
As a device manufacturer, Nokia wants to maximize the reachability for emergency call services. When the mobile VoIP emergency standardization, regulation, deployments, and networks evolve and stabilize, the current dual mode handset logic is assumed to be updated in accordance with them.
Unfortunately not. Call barring can only be done by the network.
No. Missed, received, and dialed calls are separate logs of call history. This same UI concept is used for all Nokia mobile devices.
If the registration is in manual mode, all connections, WLAN and SIP, will be disconnected. In automatic mode, the Nokia N80 Internet Edition registers back to the SIP account when the WLAN coverage is again reached.
A common reason for a registration failure is that the registrar or, for example, the SIP application layer gateway (SIP-ALG) on the route modifies the Contact URI of the response.
If the Contact URI does not match the original value in the REGISTER request, that is, if it is modified, the Nokia N80 Internet Edition checks if the hostname part matches the address in the received parameter of the Via header and the rport value. If these do not match, or if any URI parameter, such as transport=UDP, is removed or changed, the registration is treated as failed.
According to Nokia’s interpretation of the SIP RFC 3261, the registrar shall not modify the Contact URI sent in the response to the register request. When forwarding a request, for example, INVITE, some SIP servers write their own address to the Contact URI, instead of correctly inserting a Record-Route header. This also works with the Nokia N80 Internet Edition, but as for the REGISTER request the Contact has a different meaning and should not be touched.
There have been conflicting opinions about how features available in both GSM and voice-over-IP (VoIP) technologies should be implemented. Nokia will study the issue to see if those settings could somehow be combined in the future.
Unfortunately not in the current software release, but a swap between two calls is possible.
No, each realm can have several profiles but the user name must be the same in all of them.
Yes. When you use a terminal without a subscriber identity module (SIM) card, an offline profile is in use. With this profile, the terminal can be used in places where GSM phones are not allowed, such as airplanes. The WLAN connection might not be allowed in those circumstances either, and therefore you will be requested to confirm the creation of a WLAN connection in offline mode.
No. When the hang-up button is pressed, the held call is disconnected. To clear the screen, use the "C" button. All Nokia mobile devices work in a similar way.