Tieline Bridge-IT, User Manual

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© Tieline Research Pty. Ltd. 2019
Bridge-IT Manual v4.0
Connecting a Multicast Client Program
1. After you have created multicast server and client programs on your codecs you can dial
multicast connections. First select the multicast server program you want to use on the server
codec and dial to connect.
2. Select and load the multicast client program on each of the multicast client codecs and dial the
multicast IP address to begin receiving multicast audio packets.
a. Press the HOME button to return to the Home screen.
b. Use the navigation buttons to select Programs and press the button.
c. Use the up and down navigation buttons to select the multicast client program you
want to connect with, then press the button to load the program.
d. Press the CONNECT button to make a connection.
Navigate to Cxns on the Home screen to view a codec's connection Status , then press to view
connection statistics for IP packets being received over the connection.
13.11 About SIP
SIP provides interoperability between different brands of codecs due to its standardized protocols
for connecting different devices. The codec is fully EBU N/ACIP Tech 3326 compliant when
connecting using SIP (Session Initiation Protocol) to other brands of IP codecs.
SIP is also a useful way of dialing another device and locating it easily. This task is usually
performed by SIP servers, which communicate between SIP-compliant devices to set up a call.
SIP connections can be made in two ways; registered or unregistered.
Unregistered Peer-to-Peer SIP Connections
Codecs don’t need to be registered to a SIP server to dial peer-to-peer SIP connections. An
unregistered SIP peer-to-peer connection involves two codecs connecting to each other directly
using an IP address, as you would for a standard Tieline IP call. The difference is that a Tieline IP
call uses proprietary Tieline session data to negotiate call parameters (e.g. algorithm and bit rate)
when a call is established, whereas a peer-to-peer SIP connection uses Session Description
Protocol (SDP) for this purpose. SIP provides interoperability between different brands of codecs
due to its standardized protocols for connecting dissimilar devices and is used when connecting
Tieline codecs to non-Tieline devices.
There are two very distinct parts to a call when dialing over IP. The initial stage is the call setup
stage and this is what SIP and SDP is used for. The second stage is when data transfer occurs
and this is left to the other protocols such as RTP/UDP to stream audio data. SDP works with a
number of other protocols, to deliver the following functions when connecting devices over SIP:
· Establish a codec’s location.
· Determine the availability of a codec.
·
Negotiate the features to be used during a call, e.g. the algorithm and bit rate.
·
Provide call management of participants.