The codec is also capable of creating multiple SIP Peer-to-Peer connections. Configure a new program and configure each SIP audio stream as you would for a single SIP Peer-to-Peer program.
The SIP UDP audio ports are automatically allocated by the Java Toolbox Web-GUI when you create SIP programs incorporating multiple audio streams. The first stream uses UDP audio port 5004 and then each subsequent stream created will in the following order use UDP audio ports 5006, 5008, 5010, 5012 and 5014. These audio ports need to be open in your firewall at each end of the connection to allow the successful transfer of audio packets.
To answer multiple SIP calls you need to create and lock a suitable SIP answering program in the codec, or it will be unloaded by the first SIP call and a default peer-to-peer program will be loaded. There is no Tieline session data transferred during SIP calls to assist with configuring the codec.
Important Notes: •Remember to lock an answering program in a codec when answering multiple SIP calls. •When multiple calls are answered by the codec they are routed to audio inputs and outputs on a first come, first served basis. •Ensure the appropriate UDP audio ports are open in your firewall to allow multiple SIP audio streams to connect. See Installing the Codec at the Studio for more information. •Failover and SmartStream PLUS redundant streaming are not available with SIP connections. |