Codec Tieline Gateway User Manual page 51

Some SIP servers route RTP audio through the SIP server as well and Tieline recommends avoiding
this type of server whenever possible. Otherwise you will be reliant on the SIP server for streaming
broadcast audio packets and most servers are not designed for mission critical packet streaming.
To dial a codec via a SIP server requires:
1. Both devices to be registered with separate SIP accounts.
2. Both codecs configured to operate in SIP mode.
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 ports
5004 to 5054.
A SIP server administrator should be able to provide the following details to enable SIP registration
of a device:
· Username
· Authorized User
· 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 Dialing SIP Peer-to-Peer. 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.
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.
© Tieline Research Pty. Ltd. 2021
Gateway and Gateway 4 Manual v1.2
51