Problems when trying to register with a SIP provider
Let’s assume that the WLAN and IP level connectivity has been verified with the browser and you encounter problems when trying to register with a SIP provider, then check the issues hereunder.
You need a SIP client configuration with service provider-specific parameters. If you received the configuration from the service provider, you only need to configure the user name/number and password (detail you’ve got from your VoIP service provider). Let’s say the Web browser works but the SIP registration fails, then check:
- Is the account yet/still valid and can you register using any other SIP client (such as a PC)? Some service providers require you to open a special Web page to activate the account. A link to the page is typically sent in an e-mail.
- If you had to fill in the user name and password manually, double check the user name and re- enter the password. Remember that they are case sensitive.
You can check your the registration status from
Menu -> Settings -> Connection -> SIP settings, or the active connections from Menu -> Connectivity -> Conn. mana… -> Active data connections.
If you are making the configuration manually or if you are working for a service provider preparing the configuration to be sent over the air to the end users, please note:
- Do not enable the advanced IMS network features, such as IMS SIP dialect mode (default is IETF), security negotiation, or signaling compression, if your service platform does not support them.
- Disabling loose routing is extremely seldom required. Strict routing is disapproved.
- Make sure that the user name for the registrar and proxy authentication is in correct format. The authentication user name may simply be the user name part of the public SIP URI, but it may also contain the SIP domain, or be totally different, that is, a private ID used only for the authentication.
- The authentication parameter values in the proxy settings are typically identical to those defined for the registrar and can therefore be left out. However, the values are needed if the proxy and registrar require different credentials.
- The registrar’s port is almost always the default value of 5060, but in some cases the SIP proxy or Session Border Controller (SBC) may use a non-standard port value, for example, 5070.
- Check whether the proxy accepts or rejects the REGISTER request if it contains a Route header.
- If the proxy accepts requests with the Route header, configure the proxy address into the SIP settings.
- If the proxy rejects requests with the Route header, do not configure the proxy address into the SIP settings.
- Use the following guidelines to select the Transport parameter value either as ‘Auto’, ‘TCP’, or ‘UDP’ for the SIP settings of the first hop (which for REGISTER request is the proxy, if such is configured, but otherwise the registrar):
- The ‘Auto’ mode in the settings selects the UDP or TCP transport dynamically according to the message size as defined in RFC 3261. This mode should be used if the first hop is fully RFC 3261 compatible and there is no Network Address Translator (NAT) or firewall in the route (such as inside corporate intranets).
- In the ‘Auto’ mode, the VoIP client on a Nokia S60 device attempts to detect the transport selection based on the DNS records obtained. For example, if the DNS SRV record only exists for the UDP protocol, TCP is not used.
- On the other hand, in the public Internet using the’ Auto’ mode may yield to incoming calls lost because only the UDP port is kept bound to the NAT and the proxy may try to use TCP.
- If the first hop supports persistent TCP transport, select ‘TCP’ in its settings and enable the CRLF keep-alives in the advanced settings.
- If the persistent TCP is not supported or using it is not recommended by the service provider, select ‘UDP’.
- If the proxy is configured, set the Transport parameter of the registrar to value ‘Auto’.
- The STUN server address can only be set manually in the Advanced Settings of the SIP VoIP Settings application. The address can be obtained from the DNS SRV record, unless using STUN is disabled by setting the address to 0.0.0.0. The client attempts to obtain the shared secret from the STUN server over TLS, unless the DNS SRV record only exists for the UDP transport. If the TLS connection fails, the client falls back to using a normal unprotected STUN request over UDP.
No Internet call services available?
If the message ‘Internet call service not available’ is displayed when you try to establish an Internet call from the phone book’s Internet telephony field, no valid Internet telephone profile is configured or the SIP settings have been set incorrectly.
Check Menu > Connectivity > Internet co… for the configured VoIP profiles and the SIP profiles they use. Note that the path may vary according to the device.
Are you unable to connect to the VoIP service?
Check that the terminal is in the coverage of a WLAN network and that it has enough time to find a network. Note that scanning of WLAN networks is not continuous but takes place in every few minutes, depending on the WLAN configuration of the device.
If you are unable to connect to the VoIP service when you try to register into a VoIP service either from the phone book’s VoIP service tab or when making an Internet call, the registering with a SIP provider has failed. Although the AP has been successfully created, some of the settings may have been incorrectly configured or there is no connection to the proxy. For more details on the SIP profile settings, see earlier blogs.
Check the SIP profile settings as follows:
- Check that the authentication user name and password stored in the SIP settings are correct and match the service provider.
- Check that the domain names or IP addresses configured to the proxy and/or registrar are correct and the DNS server is correctly configured.
- Check that the security negotiation is disabled if the proxy does not support the procedures specified in RFC 3329.
If there is a NAT between the terminal and the proxy, the service provider may have a STUN server from which the terminal obtains its public IP address used in the Contact header. (For more information on the NAT and STUN, see http://en.wikipedia.org/wiki/Network_address_translation and http://en.wikipedia.org/wiki/STUN respectively.)
- If the domain’s DNS server is not configured with a SRV record for STUN service (the domain names of the SIP and STUN server being the same), the STUN server address must be separately configured to the device.
- Some SIP servers can handle the NAT traversal without an external STUN server.
Note: The STUN protocol cannot be used with symmetric NATs which provide address and port- dependent mapping of the allocated public IP address. The only exception is that the STUN server is embedded in the SIP server .
- Check if the NAT or firewall between the terminal and the proxy has SIP Application Layer Gateway (ALG) function enabled. If yes, retry the registration after disabling the SIP ALG function.In some NAT or firewall implementations, a malfunctioning SIP ALG may cause a registration failure.
If there is a long time-out before the ‘Unable to connect to the connection network’ error notification appears, the most probable reason is that the SIP REGISTER or response to it is lost.
If the ‘Unable to connect to the connection network’ notification appears almost instantly, the reason is that the VoIP client cannot resolve the proxy/registrar address, there is a response that is not accepted (check that the authentication user name and password are defined at least in the registrar settings), or the registrar does not accept the authentication (an invalid user name or password, the account is not yet active).
Note that all menu examples should be considered as examples only. The menu contents or the location of applications and folders may vary according to the device.