Configuring Sip Interfaces - Codec Tieline ViA User Manual

164 ViA User Manual v3
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 various 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.
· Tieline supports RFC5109 and RFC2733 compliant FEC over SIP from firmware
v2.18.xx.
· 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.
30.1

Configuring SIP Interfaces

The codec supports dialing over two SIP interfaces simultaneously.
Important Notes:
1. SIP interfaces are disabled by default.
2. SIP1 is configured to use LAN1 by default, which is mapped to the Primary Via
3. SIP2 is configure to use Wi-Fi by default, which is mapped to the Tertiary Via
4. SIP1 and SIP2 each need to use a separate IP interface when connecting, e.g. LAN1
5. SIP1 and SIP2 can however each make multiple SIP calls, e.g. two calls can be
6. The settings for SIP1 and SIP2 cannot be edited if the interface is enabled.
7. Enter a public IP address in the Public IP menu if you want to dial over SIP from
To configure SIP1 or SIP2:
1. Press the HOME
2. Tap SIP to expand the menu and then tap Interfaces.
 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]
interface by default.
interface by default.
or LAN2.
made over SIP1, or two calls can be made over SIP2.
behind a firewall. Then configure port forwarding to route traffic to the codec's local IP
address behind your firewall. Note: Do not enter a Public IP address if STUN is
configured. They cannot be used together because both will attempt to use a public IP
address over SIP. STUN settings are prioritized and used if both are configured.
button to return to the Home screen and tap Settings
> General > Tieline
.
© Tieline Research Pty. Ltd. 2020