Answering Multiple Sip Peer-To-Peer Calls - Codec Tieline Gateway User Manual

172 Gateway and Gateway 4 Manual v1.2

Answering Multiple SIP Peer-to-Peer Calls

23.19
The codec is capable of accepting multiple SIP Peer-to-Peer connections and routing them in a
deterministic manner to specific audio input and outputs. Tieline session data, in conjunction with
unique audio ports, normally facilitates routing of multiple incoming streams to specific inputs and
outputs on a single answering codec. Session Description Protocol (SDP) is used over SIP instead
of Tieline session data, therefore deterministic routing of incoming calls needs to be done a bit
differently. There are two options:
1. SIP Accounts: Using the SIP Accounts panel in the HTML5 Toolbox web-GUI, register up
to 16 SIP accounts and configure a unique answer Route in each SIP account. Then create
a multiple stream answering program (e.g. 2 x Mono Peer-to-Peer) and configure an
answer route in each audio stream matching the routes used for each SIP account you have
registered to the codec. In this way, when a call is received via a particular account, the call
is routed to the audio stream with the matching audio answer route. This process also
reliably predetermines which inputs and outputs are used on the answering codec (e.g.
Input 1 and Output 1 by default for the first audio stream in a multiple mono stream
program).
2. SIP Interfaces: When answering multiple incoming peer-to-peer SIP calls, configure an
answer Route for incoming peer-to-peer calls via the SIP Interface panel and route them to
specific audio streams. Configure answer routes in Interface SIP1 and Interface SIP2 to
match answer routes configured in an answering program's audio streams. This process also
reliably predetermines which inputs and outputs are used on the answering codec (e.g.
Input 2 and Output 2 by default for the second audio stream in a multiple mono stream
program).
© Tieline Research Pty. Ltd. 2021