Administrator’s Guide for Akuvox SP-R5xP IP Phones
Akuvox Proprietary and Confidential. Copyright © 2014 Akuvox network Co., Ltd..
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In typical commercial IP telephony deployments, all calls go through a SIP proxy server. The
requesting phone is called the SIP user agent server (UAS), while the receiving 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 is routed back to the UAS, and a direct
peer-to-peer session is established between the two UAs. Voice traffic is transmitted between
UAs over dynamically-assigned ports using Real-time Protocol (RTP).
RTP transmits real-time data such as audio and video; it does not guarantee realtime delivery
of data. RTP provides mechanisms for the sending and receiving applications to support
streaming data. Typically, RTP runs on top of UDP.
Server redundancy
Server redundancy is often required in VoIP deployments to ensure continuity of phone service,
for events where the server needs to be taken offline and the connect fails.
Two types of redundancy are possible. In some cases, a combination of the two may be
deployed:
Failover: In this mode, the full phone system functionality is preserved by having a second
equivalent capability call server take over from the one that has gone down/off-line. This
mode of operation should be done using the DNS mechanism from the primary to the
secondary server.
Fallback: In this mode, a second less featured call server with SIP capability takes over call
control to provide basic calling capability, but without some advanced features offered by
the working server (for example, shared line, call recording and MWI). IP phones support
configuration of two SIP servers per SIP registration for fallback purpose.
