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. LAN1 or LAN2.
3.Create a SIP program using the HTML5 Toolbox Web-GUI.
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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. •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 is not available with SIP and SmartStream PLUS redundant streaming is not available with SIP or sessionless visit www.tieline.com. •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] |
