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Genie Distribution WNET Manual v3
© Tieline Research Pty. Ltd. 2020
20.23
Configuring SIP
The codec is fully EBU N/ACIP Tech 3326 compliant when connecting using SIP (Session Initiation
Protocol) to other brands of IP codecs. For more background on SIP connections and the
differences between registered and unregistered peer-to-peer SIP connections see
.
To configure the codec to dial over SIP using a SIP Server you will need to:
1. Register the codec to a SIP server using SIP account credentials.
2. Configure a SIP interface in the codec. Note: This
SIP1
or
SIP2
interface will include the
proxy and port settings, as well as the selected IP interface used to make the connection,
e.g.
ETH1
or
ETH2
.
3.
Home screen
, or create a SIP program
using the HTML5 Toolbox Web-GUI.
Important Notes:
·
The codec supports dialing over SIP using a registered SIP server account, or peer-to-
peer using one of the two SIP interfaces
SIP1
and
SIP 2
.
·
SIP dialing is only supported over point-to-point connections.
·
Some ISPs and/or cellular networks may block SIP traffic over UDP port 5060.
·
Tieline G3 codecs do not support connections using algorithms like AAC, aptX
Enhanced and Opus and will default to MPEG Layer 2 if an incoming call is configured
to use these algorithms.
·
Failover is not available with SIP and SmartStream PLUS redundant streaming is not
available with SIP or sessionless visit
.
·
When connecting to a Tieline G3 codec using SIP you need to manually select the G3
audio reference level. To do this select
Home screen
> Settings > Audio >
Reference Level > Tieline G3
. In addition, select the following on the G3 codec prior
to dialing.
Select either a mono or stereo profile
Select
[Menu] > [Configuration] > [IP1 Setup] > [Session Type] > [SIP]
Select
[Menu] > [Configuration] > [IP1 Setup] > [Algorithm] > [G711/G722 or
MP2]