In typical commercial IP telephony deployments, all calls go through a SIP Proxy Server. The receiving phone
is called the SIP user agent server (UAS), while the requesting phone is called the user agent client (UAC).
SIP message routing is dynamic. If a SIP proxy receives a request from a UAS for a connection but cannot
locate the UAC, the proxy forwards the message to another SIP proxy in the network. When the UAC is
located, the response routes back to the UAS, and the two UAs connect using a direct peer-to-peer session.
Voice traffic transmits between UAs over dynamically assigned ports using Real-time Protocol (RTP).
RTP transmits real-time data such as audio and video; RTP does not guarantee real-time delivery of data. RTP
provides mechanisms for the sending and receiving applications to support streaming data. Typically, RTP
runs on top of UDP.
SIP Over TCP
To guarantee state-oriented communications, the Cisco IP 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 of UDP port-blocking by corporate firewalls. With TCP, new ports do not need
to be open 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 agent serves different domains, for example,
you do 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 IP 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 hostnames, priority, listening ports,
and so forth. The Cisco IP Phone tries to contact the hosts in the order of their priority.
If the Cisco IP 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 through primary proxy
first. If there is no response for the new INVITE from the primary proxy, after timeout, the phone attempts
to connect with the alternate proxy. If the phone fails to register to the primary proxy, it sends an INVITE to
the alternate proxy without trying the primary proxy.
Dual registration is supported on a per-line basis. Three added parameters can be configured through web
user interface and remote provisioning:
• Alternate Proxy—Default is empty.
• Alternate Outbound Proxy—Default is empty.
• Dual Registration—Default is NO (turned off).
Cisco IP Phone 8800 Series Multiplatform Phone Administration Guide for Release 11.3(1) and Later
453
Technical Details
SIP Over TCP
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