Objective
Session Initiation Protocol (SIP) is a signaling protocol which is used for managing communication sessions like voice and video calls over the Internet Protocol (IP). The protocol can be used to create, modify and terminate unicast or multicast sessions. One or more media streams may be present in the sessions.
The applications of the SIP include video conferencing, streaming multimedia distribution, file transfer and so on. SIP is an application layer protocol independent of the Transport layer below.
The objective of this document is to explain the configuration of SIP Settings for Extensions on SPA9X2 Series IP Phones.
Applicable Devices
• SPA9X2 Series IP Phones
Software Version
• v6.1.5a
Configuration of SIP Settings for Extensions in Basic Mode
Step 1. Log in to the web configuration utility and click Admin > Login > Basic > Ext1 for Basic Mode.
Step 2. Enter the SIP port number of the SIP message listening and transmission port in the SIP Port field. The default value is 5060.
Step 3. Choose an option from the SIP Debug Option drop down list. The SIP messages are sent from or received at the proxy listen port. SIP Debug feature checks the SIP messages which are to be logged. There are many options available for the SIP Debug feature. The default is None
• None — No logging.
• 1-line — Logs the Start-line only for all messages.
• 1-line excl. OPT — Logs the Start-line only for all messages except OPTIONS request or responses.
• 1-line excl. NTFY — Logs the Start-line only for all messages except NOTIFY request or responses.
• 1-line excl. REG — Logs the Start-line only for all messages except REGISTER request or responses.
• 1-line excl. OPT|NTFY|REG — Logs the Start-line only for all messages except OPTIONS, REGISTER and NOTIFY request or responses.
• full — Logs all SIP messages in full text.
• full excl. OPT — Logs all SIP messages in full text except OPTIONS request or responses.
• full excl. NTFY — Logs all SIP messages in full text except NOTIFY request or responses.
• full excl. REG — Logs all SIP messages in full text except REGISTER request or responses.
• full excl. OPT|NTFY|REG — Logs all SIP messages in full text except OPTIONS, NOTIFY and REGISTER request or responses.
Configuration of SIP Settings for Extensions in Advanced Mode
Step 1. Log in to the web configuration utility and click Admin > Login > Advanced > Ext1 for Advanced Mode.
Step 2. Choose a SIP transport option from the SIP Transport drop-down list. The options available are TCP, UDP or TLS. The default is UDP. The transport option in SIP permits the configuration of what transport type a peer will accept for inbound and outbound calls.
Step 3. Enter a SIP port number of the SIP message listening and transmission port in the SIP Port field. The default value is 5060.
Step 4. Choose Yes to enable SIP 100REL from the SIP 100REL drop-down list. 100REL is a SIP provisional message reliability. This feature overcomes interoperability which is caused due to inconsistent support for SIP reliable provisional responses which is faced when the Session Border Controller (SBC) operates with different SIP networks. SBC is a device usually deployed in VoIP network to exert control over signalling and media streams. The provisional responses (1xx) in SIP do not have an acknowledgement system so they are not reliable. But in some cases these provisional responses need reliability, the use of Provisional Response Acknowledgement (PRACK) enables reliability to be offered to provisional messages. 100REL option is used to show that the reliable provisional responses are supported or required and the PRACK is used to give an ACK for reliable provisional response. The default is No.
Step 5. Enter the external SIP port in the EXT SIP Port field. These are the ports that are used for external user access.
Step 6. Choose Yes from the Auth Resync Reboot drop-down list. Resync option is configured in SIP through a SIP NOTIFY message. If this is enabled the SPA9X2 can authenticate a sender when it receives a SIP NOTIFY resync reboot message (RFC 2617). If this feature is not needed choose No. The default is Yes.
Step 7. Enter the appropriate header in the SIP Proxy-Require field. The SIP Proxy-Require is used to indicate proxy sensitive features that must be supported by proxy. The SIP proxy can support a specific extension or parameter when this header is given by the user agent. If proxy does not support this field even after configuration, then a message is sent which says unsupported.
Step 8. Choose Yes from the SIP Remote Party-ID drop-down list. This allows to use Remote Party-ID header instead of a From header. If this feature is not needed, choose No. The SIP Remote Party-ID header identifies the calling party and it includes user party, screen and privacy headers that indicate how a call is presented or screened. The default is Yes.
Step 9. Enter the appropriate period of time in seconds in the Referor Bye Delay field. This checks when the SPA9X2 sends BYE to end old call legs once the call transfers have been completed. There are many delay settings for doing this function. They are Referor, Refer Target, Referee and Refer-To-Target. The default value for Referor Bye Delay is 4.
Step 10. Choose Yes in the Refer-To-Target Contact drop-down list. This allows contacting the Refer-To-Target. If this feature is not needed, choose No. The default is No.
Step 11. Enter the appropriate period of time in seconds in the Referee Bye Delay field. The default is 0.
Step 12. Choose an option from the SIP Debug Option drop-down list. The SIP messages are sent from or received at the proxy listen port. SIP Debug feature checks the SIP messages which are to be logged. There are many options available for SIP Debug feature. The default is None
• None — No logging.
• 1-line — Logs the Start-line only for all messages.
• 1-line excl. OPT — Logs the Start-line only for all messages except OPTIONS request or responses.
• 1-line excl. NTFY — Logs the Start-line only for all messages except NOTIFY request or responses.
• 1-line excl. REG — Logs the Start-line only for all messages except REGISTER request or responses.
• 1-line excl. OPT|NTFY|REG — Logs the Start-line only for all messages except OPTIONS, REGISTER and NOTIFY request or responses.
• full — Logs all SIP messages in full text.
• full excl. OPT — Logs all SIP messages in full text except OPTIONS request or responses.
• full excl. NTFY — Logs all SIP messages in full text except NOTIFY request or responses.
• full excl. REG — Logs all SIP messages in full text except REGISTER request or responses.
• full excl. OPT|NTFY|REG — Logs all SIP messages in full text except OPTIONS, NOTIFY and REGISTER request or responses.
Step 13. Enter the appropriate period of time in seconds in the Refer Target Bye Delay field. The default is 0.
Step 14. Choose Yes from the Sticky 183 drop-down list. 180 and 183 are two types of SIP responses. 180 SIP response is a ringing message which tries to alert the user after receiving an INVITE. 183 SIP response is a session progress message which gives information about the call process which is not sorted. If Sticky 183 is enabled then the IP telephony leaves out any more 180 SIP responses after it receives its first 183 SIP response for an outbound INVITE. If this feature is not needed, choose No. The default is No.
Step 15. Choose Yes from the Auth INVITE drop-down list. This makes authorization essential for initial incoming INVITE requests from the SIP Proxy. The default is No.
Step 16. Choose Yes from the Nfty Refer On 1xx-To-Inv drop-down list. This will make the Transferee to send a NOTIFY message with Event: Refer to the transferor on the transfer call log for any 1xx response which is received from the transfer target. If No is chosen the phone will only send a NOTIFY for final responses (200 and higher).
Step 17. Choose Yes from the Use Anonymous with RPID drop-down list. This makes them FROM header's display name and user ID to be set to anonymous when the user blocks his caller ID. If No is chosen the FROM header's display name and user ID are not protected. The Remote Party-ID header displays privacy = full when the caller wishes to block the caller ID. This parameter is configured only if the SIP Remote Party-ID is set to Yes, if not this is ignored. The default for this parameter is Yes.
Step 18. Choose any option from the Set G729 annexb drop-down list. This is used to configure G729 annexb settings. G729 annexb is used to monitor the voice activity in the signal. The default is None.
Step 19. Click Submit All Changes and the changes are configured.