Voice-over-IP (VoIP) over Satellite
The ability to transmit voice calls which have been traditionally handled by the public switched telephone network (PSTN) over a lower cost IP infrastructure has become a reality. VoIP based telephony has also provided a very cost-effective solution for companies that have locations in many disparate regions and require connectivity back to their headquarters.
When engineering a satellite network capable of effectively handling VoIP, there are several important elements that require consideration:
- Latency: There is approximately 280 ms of one-way propagation delay due to the location of the Geo Stationary orbit and the speed of light. Regardless of the satellite product, this propagation delay must be considered and overcome. Round trip delays in a TDMA satellite networks are around 700 ms.
- Jitter: Packets transmitted at equal intervals from the transmitting gateway arrive at the receiving gateway at irregular intervals. Excessive jitter has the effect of making speech choppy and difficult to understand. Jitter buffers (packet buffers that hold incoming packets for a specified amount of time) are used to counteract the effects of network fluctuations and create a smooth packet flow at the receiving end. However the buffers add to latency.
- Packet Loss: Packet corruption will cause degradation of voice quality. Since all of the voice traffic is UDP/IP and can not be retransmitted - unlike in the case of TCP/IP - the information is completely lost if the packet becomes corrupted. Therefore it is important to have a satellite connection with very low Bit Error Rates (BER) to ensure low or no loss.
- QoS and Traffic Prioritization: Packet switched networks are subject to congestion as typical data traffic is bursty. Congested networks can wreak havoc on a VoIP call with delayed , dropped , or packets that are out of sequence . It is a necessity to have QoS and Prioritization in order to guarantee delivery of VoIP traffic through congested links.
- Compression Technologies: The most dominant encoding standard for narrow-band links is the G.729 codec. G.729 encoding requires 8 kbps of bandwidth, but because of the overhead associated with IP/UDP/RTP headers, the actual bandwidth needed is 24 kbps. With Compress RTP (cRTP) the total bandwidth requirement per call will drop to about 11 kbps.
- Overall bandwidth requirement: To design a network properly, one needs to know the amount of bandwidth required per VoIP call, the number of concurrent calls, and the duration of the call. As satellite bandwidth tends to be expensive, investment in sophisticated VoIP gateways which minimize bandwidth requirements amortizes quickly.