Cisco SPA 525G2 Administration Manual page 110

Configuring SIP, SPCP, and NAT
Configuring SIP
Cisco Small Business SPA 300 Series, SPA 500 Series, and WIP310 IP Phone Administration Guide
• RTP Port Max—Maximum port number for RTP transmission and reception.
<RTP Port Min> and <RTP Port Max> 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. 1 6. 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
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
- NAME is set to
- 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
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.
• 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.
Interval for sending out RTCP sender reports on an
User ID@Proxy
Display Name Anonymous
(or
Call Round Trip Delay
4
if user blocks caller ID)
value in the Info
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