Cisco SPA301 Administration Manual page 127

Configuring SIP, SPCP, and NAT
Configuring SIP
Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide
—
• RTCP Tx Interval
Protocol (RTCP) sender reports on an active connection. During an active
connection, the Cisco SPA IP phones send out compound RTCP packets.
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 one receiver
report (RR); the last SR carries no RR.
The SDES contains CNAME, NAME, and TOOL identifiers:
User ID@Proxy
- CNAME—
Display Name
- NAME—
- TOOL—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 local
time, not the time reported by an NTP server.
If the Cisco IP phone receives a RR from a peer, it tries to compute the round
trip delay and show it as the
of the phone web user interface administration page. The range is from 0 to
255 seconds. Defaults to 0.
• No UDP Checksum—Select yes to enable the Cisco IP phone to calculate
the UDP header checksum for SIP messages. Since this adds to the
computation load, we recommend the default value, no (disabled).
• Symmetric RTP—Select yes to enable Symmetric RTP operation. When
enabled, it sends RTP packets to the source address and the port of the last
received valid inbound RTP packet. If disabled (or before the first RTP
packet arrives) it sends RTP to the destination as indicated in the inbound
SDP. Defaults to no.
• Stats in BYE—Select yes to send the P-RTP-Stat header in response to a
BYE message. The header contains the RTP statistics on the current call.
Defaults to no.
The format of the P-RTP-Stat header is:
P-RTP-State: PS=<packets sent>,OS=<octets
sent>,PR=<packets received>,OR=<octets
received>,PL=<packets lost>,JI=<jitter in ms>,LA=<delay
in ms>,DU=<call duration in s>,EN=<encoder>,DE=<decoder>
Interval for sending out Real-Time Transport Control
Anonymous
(or
Call Round Trip Delay
if user blocks caller ID)
value in the Info section
4
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