Sip Over Tcp; Sip Proxy Redundancy; Dual Registration - Cisco 8831 Administration Manual

SIP and Cisco Unified IP Conference Phone 8831 for Third-Party Call Control

SIP Over TCP

To guarantee state-oriented communications, Cisco conference phone can use TCP as the transport
protocol for SIP. This protocol provides guaranteed delivery that assures that lost packets are
retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were
sent.
TCP overcomes the problem UDP ports have of being blocked by corporate firewalls. With TCP, new
ports do not need to be opened or packets dropped, because TCP is already in use for basic activities,
such as Internet browsing or e-commerce.

SIP Proxy Redundancy

An average SIP proxy server can handle tens of thousands of subscribers. A backup server allows an
active server to be temporarily switched out for maintenance. Cisco phones support the use of backup
SIP proxy servers to minimize or eliminate service disruption.
A static list of proxy servers is not always adequate. If your user agents are served by different domains,
for example, you would not want to configure a static list of proxy servers for each domain into every
Cisco IP phone.
A simple way to support proxy redundancy is to configure a SIP proxy server in the Cisco conference
phone configuration profile. The DNS SRV records instruct the phones to contact a SIP proxy server in
a domain named in SIP messages. The phone consults the DNS server. If configured, the DNS server
returns an SRV record that contains a list of SIP proxy servers for the domain, with their host names,
priority, listening ports, and so forth. The Cisco conference phone tries to contact the hosts in the order
of their priority.
If the Cisco conference phone currently uses a lower-priority proxy server, the phone periodically probes
the higher-priority proxy and switches to the higher-priority proxy when available.

Dual Registration

The phone always registers to both primary (or primary outbound) and alternate (or alternate outbound)
proxies. After registration, the phone sends out Invite and Non-Invite SIP messages via primary proxy
first. If there is no response for the new INVITE from the primary proxy, after timeout, the phone should
attempt with the alternate proxy.
Dual registration is supported per line basis. Three new parameters are added which can be configured
via Web GUI and remote provisioning:
•
•
•
Upon configuring the parameters, reboot the phone for the feature to take effect.
The administrator should specify a value for primary proxy (or primary outbound proxy) and alternate
Note
proxy (or alternate outbound proxy) for the feature to function properly.
Cisco Unified IP Conference Phone 8831 for Third-Party Call Control Administration Guide
3-2
Alternate Proxy—Default is empty
Alternate Outbound Proxy—Default is empty
Dual Registration—Default is NO (turned off)
Chapter 3 Configure SIP and NAT