Tieline ViA User Manual page 129

Codec
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3. The IP address of the SIP server.
4. An IT administrator to open UDP port 5060 to enable SIP traffic, as well as UDP audio port
5004.
A SIP server administrator should be able to provide the following details to enable SIP registration
of a device:
·
Authenticating username
·
SIP address
·
Domain
·
Realm
·
Registrar
·
Registar port
·
Outbound Proxy
·
Proxy port
Getting Started with SIP
To dial over SIP peer-to-peer without using a SIP server see
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. LAN1 or LAN2.
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.
·
The codec supports a SIP call being placed on-hold. Note: there are several different
implementations of "on-hold" by different SIP providers. Some will stop streaming when
a call is placed on-hold and others will replace live streaming with on-hold messages or
music.
·
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 Audio
G3. In addition, select the following on the G3 codec prior to dialing.
© Tieline Pty. Ltd. 2017
§
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]
ViA User Manual v2.0
Dialing SIP
Peer-to-Peer. To dial over
> General
129
> Tieline

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