Linksys WIP310 SPA500 Series and WIP310 IP Phone Administration Guide - Page 230

carries no RR. The SDES contains CNAME, NAME, and, No UDP Checksum

Page 230 highlights

Cisco SPA 500 Series and Wireless IP Phone Field Reference SIP Tab B Parameter RTCP Tx Interval No UDP Checksum Symmetric RTP Description Interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds. During an active connection, the IP phone can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a SR (Sender Report) and a SDES (Source Description). The last RTCP packet contains an additional BYE packet. Each SR except the last one contains exactly 1 RR (Receiver Report); the last SR carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers. The CNAME is set to @, NAME is set to (or Anonymous if user blocks caller ID), and TOOL is set to the Vendor/Hardware-platform-software-version (such as Cisco/IP phone-1.0.31(b)). The NTP timestamp used in the SR is a snapshot of the IP phone's local time, not the time reported by an NTP server. If the IP phone receives a RR from the peer, it attempts to compute the round trip delay and show it as the value (ms) in the Info section of IP phone web page. Defaults to 0. Select yes if you want the IP phone to calculate the UDP header checksum for SIP messages. Otherwise, select no. Defaults to no. Enable symmetric RTP operation. If enabled, sends RTP packets to the source address and port of the last received valid inbound RTP packet. If disabled (or before the first RTP packet arrives) sends RTP to the destination as indicated in the inbound SDP. Defaults to no. Cisco SPA 500 Series and WIP310 IP Phone Administration Guide 219

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Cisco SPA 500 Series and Wireless IP Phone Field Reference
SIP Tab
Cisco SPA 500 Series and WIP310 IP Phone Administration Guide
219
B
RTCP Tx Interval
Interval for sending out RTCP sender reports on an
active connection. It can range from 0 to 255 seconds.
During an active connection, the IP phone can be
programmed to send out compound RTCP packet on
the connection. Each compound RTP packet except
the last one contains a SR (Sender Report) and a SDES
(Source Description). The last RTCP packet contains an
additional BYE packet. Each SR except the last one
contains exactly 1 RR (Receiver Report); the last SR
carries no RR. The SDES contains CNAME, NAME, and
TOOL identifiers. The CNAME is set to <User
ID>@<Proxy>, NAME is set to <Display Name> (or
Anonymous if user blocks caller ID), and TOOL is set to
the Vendor/Hardware-platform-software-version (such
as Cisco/IP phone-1.0.31(b)). The NTP timestamp used
in the SR is a snapshot of the IP phone’s local time, not
the time reported by an NTP server. If the IP phone
receives a RR from the peer, it attempts to compute the
round trip delay and show it as the <Call Round Trip
Delay> value (ms) in the Info section of IP phone web
page.
Defaults to 0.
No UDP Checksum
Select yes if you want the IP phone to calculate the
UDP header checksum for SIP messages. Otherwise,
select no.
Defaults to no.
Symmetric RTP
Enable symmetric RTP operation. If enabled, sends
RTP packets to the source address and port of the last
received valid inbound RTP packet. If disabled (or
before the first RTP packet arrives) sends RTP to the
destination as indicated in the inbound SDP.
Defaults to no.
Parameter
Description