Tieline Bridge-IT, User Manual

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© Tieline Research Pty. Ltd. 2019
Bridge-IT Manual v4.0
1. Register the codec to a SIP server using SIP account credentials.
2. Configure a SIP interface in the codec. Note: The
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.
LAN1 .
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, not multi-unicast
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 and SmartStream PLUS redundant streaming are not available with SIP
connections.
· When connecting to a Tieline G3 codec using SIP you need to manually select the G3
audio reference level in the codec. To do this select
SETTINGS > Audio > Ref
Level > Tieline G3 . In addition, configure 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]