Linksys WIP310 SPA500 Series and WIP310 IP Phone Administration Guide - Page 102
User ID@Proxy, Display Name, Anonymous, Call Round Trip Delay, RTP Port Min, RTP Packet Size
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Configuring SIP, SPCP, and NAT Configuring SIP 4 • RTP Port Min-Minimum port number for RTP transmission and reception. and should define a range that contains at least 10 even number ports (twice the number of lines); for example, 100 - 106. Defaults to 16384. • RTP Port Max-Maximum port number for RTP transmission and reception. and should define a range that contains at least 10 even number ports (twice the number of lines); for example, 100 - 106. Defaults to 16482. • RTP Packet Size-Packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds. Defaults to 0.030. • Max RTP ICMP Err-Number of successive ICMP errors allowed when transmitting RTP packets to the peer before the Cisco IP phone terminates the call. If the value is set to 0 (the default), the Cisco IP phone ignores the limit on ICMP errors, disabling the feature. • RTCP Tx Interval-Interval for sending out RTCP sender reports on an active connection. During an active connection, the Cisco IP phone can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a sender report (SR) and a source description (SDES). The last RTCP packet contains an additional BYE packet. Each SR except the last one contains exactly 1 receiver report (RR); the last SR carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers.: - CNAME is set to User ID@Proxy - NAME is set to Display Name (or Anonymous if user blocks caller ID) - TOOL is set to the Vendor/Hardware-platform-software-version (such as Cisco/SPA9000-5.2.2(SCb)). - The NTP timestamp used in the SR is a snapshot of the Cisco IP phone's local time, not the time reported by an NTP server. - If the Cisco IP phone receives a RR from the peer, it tries to compute the round trip delay and show it as the Call Round Trip Delay value in the Info section of the web GUI administration page. It can range from 0 to 255 seconds. Defaults to 0 (recommended). • No UDP Checksum-Select yes if you want the Cisco IP phone to calculate the UDP header checksum for SIP messages. Since this involves computation load, you should keep the default value (no) to disable it. Cisco SPA 500 Series and WIP310 IP Phone Administration Guide 91