Tieline Merlin PLUS User Manual page 204

204 Merlin PLUS User Manual v2.0
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 and SmartStream PLUS redundant streaming are not available with SIP
connections.
·
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]
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