SIP is commonly used for internet media that is real-time such as Voice over Internet Protocol (VoIP) phone calls, video conferencing, and instant messaging. Real-time media is media that is not pre-recorded and is defined by audio, video, and text messaging that needs to be as close to instantaneous as possible.
Any delays or loss in the real-time media generally will not be re-transmitted as by the time the media packets are re-transmitted they will no longer be relevant because the “conversation” has already moved past the timeframe to which the packets were relevant. Because of this time-sensitive requirement, the UDP (User Datagram Protocol) protocol is most commonly used since there is no packet checking and retransmission as seen in TCP (Transmission Control Protocol) therefore allowing packets to be sent more quickly.
Now it is important to configure your communications infrastructure/platform.
- To get started ensure you are sending calls to trunk-production-us1.avoxi.com
- RTP port range for AVOXI: 10000-65535
AVOXI supports UDP, TCP, and TLS version 1.2 (follow here to learn more). Open the following ports to ensure proper connectivity:
- 5060 UDP
- 5060 TCP
- 5061 TLS
- Supported Codecs: G.711u(ulaw), G.711a(alaw), and G.729
Designated IPs: Below is our designated IPs that will need to be whitelisted to allow forward to SIP in AVOXI
SIP Forwarding for Inbound Forwarding
Use the below guides and quick links to configure the SIP account all calls should be diverted too:
- Add New SIP URI
SIP Trunking for Outbound Termination
Use the below quick links to view, manage and configure SIP Trunking within your AVOXI online portal.