Tieline Bridge-IT, Руководство пользователя

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124 Bridge-IT Manual v4.0
© Tieline Research Pty. Ltd. 2019
2. Click Save Program to complete configuration of the program.
3. Click Finish to exit the wizard.
4. The newly created program will be displayed in the left pane within the Program Manager
panel and in the Program Loader panel . Select and connect audio streams in a program
using the Connections panel , or dial the program manually using the codec front panel.
Note: When you initially dial using a multi-unicast program all audio streams are dialed. It
is then possible to connect and disconnect audio streams individually.
5.
16.10 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
About
SIP .
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.
LAN or VLAN .
3. Create a SIP program using the front panel via the 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; SmartStream PLUS redundant streaming is not
available with SIP or sessionless IP..
·
For detailed information about connecting with other brands of codec using SIP click
http://www.tieline.com/interoperability
· 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]