Tieline ViA User Manual page 128

128 ViA User Manual v2.0
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.
·
Adjust session management features while a call is in progress (e.g. termination and
transfer of calls).
All the mandatory EBU N/ACIP 3326 algorithms are supported in your Tieline codec, including
G.711, G.722, MPEG-1 Layer 2 and 16 bit PCM, as well as optional algorithms including Opus, LC-
AAC, AAC-LD, HE-AACv2 and aptX Enhanced. The default algorithm selected when connecting
using SIP is G.711.
SIP Server Connections
The benefit of using a SIP server to connect is that any device can be 'discovered' via its SIP server
registration. This is particularly useful if a codec is being used in multiple locations with IP
addresses that are DHCP assigned. These DHCP addresses are unreliable and are not
recommended for live broadcast connections. As long as your codec and the device you are dialing
are both registered to a SIP server you can connect by simply dialing the destination SIP address.
Some SIP servers route RTP audio through the SIP server as well and Tieline recommends avoiding
this type of server whenever possible. Otherwise you will be reliant on the SIP server for streaming
broadcast audio packets and most servers are not designed for mission critical packet streaming.
To dial a codec via a SIP server requires:
1. Both devices to be registered with separate SIP accounts.
2. Both codecs configured to operate in SIP mode.
© Tieline Pty. Ltd. 2017