AUDIO STREAMS |
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9.3 Real-time Transport Protocol
Regardless of whether you use TSCP or SIP to initiate your connection, the audio is sent using the Real-time
Transport Protocol (RTP). Rather than using a connection protocol, you may send RTP streams manually.
The features of RTP connections are:
♦
Cross-product compatibility
♦
Point-to-multipoint operation
♦
Unidirectional audio transfer
♦
Agile Connection Technology (on applicable codecs)
There are no servers to configure for RTP. To send an RTP stream to another codec, you enter its IP address and port
number in the format ww.xx.yy.zz:port or, if a hostname is available, hostname.tld:port. The remote codec must be
listening on that port for a connection, and that port must be open to the public network. On the Z/IP ONE, the
RTP listen defaults to UDP port 9150. This can be configured on the second page of the Setup->Network menu on the
front panel, or on the Network web page.
RTP streams push audio in one direction – from the dialing unit to the dialed unit. Because of this, it can be used in
point-to-multipoint operation. To use this function, just issue multiple RTP connections, one after another. The
only limit to the number of connections is your outgoing bandwidth. Using the DISC button on the front panel will
stop all outgoing RTP streams at once.
Although RTP streams are push-only, you can transfer audio in each direction. To do this, the operator of each
codec must send an RTP stream to the other. If this is done, your Z/IP ONE will have all the connection features
(including Agile Connection Technology and remote quality information) as if the transfer was started via TSCP
or SIP. However, since the call is not handled by a connection manager, each end must still stop its own stream
individually.
In some scenarios, this process can be automated. The Z/IP ONE has what we call ‘reciprocal RTP.’ If you send an
RTP stream to one of the Z/IP ONE’s reciprocal RTP ports, it will respond to the ‘caller’ with its own stream. This
sets up a bidirectional connection, just like using TSCP or SIP, but there is no NAT traversal. The called party must
have a port or ports forwarded in order for reciprocal RTP to work.
In addition to the listen-only RTP port, there are three reciprocal RTP ports. These are the next three port numbers
above the listen port, and are disabled by default (e.g. if the RTP listen port is 9150, the available reciprocal RTP
ports are 9151, 9152, and 9153). Each of these ports has a different behavior, and can be enabled individually if
needed:
Port
Default
Effect
Base
9150
Receive Only
Base + 1
9151
Reply with G.722
Base + 2
9152
Reply with same as received codec
Base + 3
9153
Reply with current codec setting
Setting the RTP base port to 0 will disable the RTP receive-only port as well as all reciprocal RTP ports.
