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The proxy uses these re-INVITE requests to maintain the status' of the connected sessions. See RFC4028 for details. 8 In the "Timer 1 and Timer 2" fields, enter a time, in milliseconds, that will apply to an IP phone session. These timers are SIP transaction layer timers defined in RFC 3261. Timer 1 is an estimate of the round-trip time (RTT). Default is 500 msec. Timer 2 represents the amount of time a non-INVITE server transaction takes to respond to a request. Default is 4 seconds. Configuring 41-001160-00, Release 2.1, Rev 04 4-61 IP Phone Administrator Guide Global SIP Settings Configuring the IP Phones Aastra Web UI Step Action 9 In the "Transaction Timer" field, enter the amount of time, in milliseconds, that the phone allows the call server (registrar/proxy) to respond to SIP messages that it sends. Valid values are 4000 to 64000. Default is 4000. Note: If the phone does not receive a response in the amount of time designated for this parameter, the phone assumes the message has timed out. 10 In the "Transport Protocol" field, select a transport protocol to use when sending SIP Real-time Transport Protocol (RTP) packets. Valid values are User Datagram Protocol (UDP) and Transmission Control Protocol (TCP), UDP, TCP, Transport Layer Security (TLS) or Persistent TLS. The value “UDP” is the default. For more information about TLS, see “RTP Encryption” on page 4-68 and Chapter 5, the section, “Transport Layer Security (TLS)” on page 6-21. 11 In the " Registration Failed Retry Timer" field, enter the amount of time, in seconds, that the phone waits between registration attempts when a registration is rejected by the registrar. Valid values are 30 to 1800. Default is 1800. 12 In the " Registration Timeout Retry Timer" field, enter the amount of time, in seconds, that the phone waits until it re-attempts to register after a REGISTER message times out. Valid values are 30 to 214748364. Default is 120. 13 In the " Registration Renewal Timer" field, enter the length of time, in seconds, prior to expiration, that the phone renews registrations. For example, if the value is set to 20, then 20 seconds before the registration is due to expire, a new REGISTER message is sent to the registrar to renew the registration. Valid values are 0 to 214748364. Default is 15. 14 The “BLF Subscription Period” field is enabled by default with a value of 3600 seconds. This feature sets the time period, in seconds, that the IP phone resubscribes the BLF subscription service after a software/firmware upgrade or after a reboot of the IP phone. For information about setting the “BLF Subscription Period”, see Chapter 5, the section, “BLF Subscription Period” on page 5-120. 15 (For Sylantro Servers) The “ACD Subscription Period” field is enabled by default with a value of 3600 seconds. This feature sets the time period, in seconds, that the IP phone resubscribes the BLF subscription service after a software/firmware upgrade or after a reboot of the IP phone. For information about setting the “ACD Subscription Period”, see Chapter 5, the section, “ACD Subscription Period” on page 5-133. 4-62 41-001160-00, Release 2.1, Rev 04 Configuring Network and Session Initiation Protocol (SIP) Features Global SIP Settings Aastra Web UI Step Action 16 (For Broadsoft Servers) The “Blacklist Duration” field is enabled by default with a value of 300 seconds (5 minutes). Valid values are 0 to 9999999. This feature sets the time period, in seconds, that the IP phone resubscribes the BLF subscription service after a software/firmware upgrade or after a reboot of the IP phone. Note: The value of “0” disables the blacklist feature. For information about setting the “Blacklist Duration”, see Chapter 6, the section, “Blacklist Duration (Broadsoft Servers)” on page 6-17. 17 Enable the "Whitelist Proxy" field by checking the check box. (Disable this field by unchecking the check box. Default is disabled). When this feature is enabled, an IP phone accepts call requests from a trusted proxy server only. The IP phone rejects any call requests from an untrusted proxy server. For information about setting the “Whitelist Proxy”, see Chapter 6, the section, “Whitelist Proxy” on page 6-19. 18 Click to save your changes. Configuring 41-001160-00, Release 2.1, Rev 04 4-63 IP Phone Administrator Guide Global SIP Settings Real-time Transport Protocol (RTP) Settings Configuring the IP Phones Real-time Transport Protocol (RTP) is used as the bearer path for voice packets sent over the IP network. Information in the RTP header tells the receiver how to reconstruct the data and describes how the bit streams are packetized (i.e. which codec is in use). Real-time Transport Control Protocol (RTCP) allows endpoints to monitor packet delivery, detect and compensate for any packet loss in the network. Session Initiation Protocol (SIP) and H.323 both use RTP and RTCP for the media stream, with User Datagram Protocol (UDP) as the transport layer encapsulation protocol. Note: If RFC2...

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