RTP Optimisation

RTP Optimisation is a technique for optimising bandwidth and quality of RTP traffic, which is particularly effective when used for VoIP over satellite-based. The technique combines four elements to produce a VoIP transport mechanism that is uniquely optimized for bandwidth efficiency and quality across Vocality networks.

Details

Vocality has introduced a technique for optimising bandwidth and quality of IP RTP traffic, which is particularly effective when using VoIP over satellite-based networks.

The technique combines four elements to produce a RTP transport mechanism that is uniquely optimized for bandwidth efficiency and quality of VoIP across Vocality networks. The first is a proprietary method of RTP compression, which by conversation-awareness strips out the constant portion of the IP Header and the UDP Header and also reduces the bandwidth used by the RTP Header well.

The second optional element increases on the bandwidth saving by explicitly stripping out any negotiation messages in the SIP session description protocol messages which request the high-bandwidth 64K G.711 codecs and forcing the connection to use 8K G.729 codecs instead.

The third optional element, filters out the small-sample packets which occur in G.729B when transitioning into and out of silence suppression mode. Many of these can occur during each second of normal speech and in trials have not been found to materially enhance or improve the voice quality; their deletion can save vital bandwidth on a satellite link.

The final element adds a timestamp to each voice packet, which is used as a means of timing the release of the packet into the remote IP network once it has transited across the satellite network or any other network, thus effectively removing jitter from the network and providing a smooth feed to the remote voice terminal. This dramatically improves the overall voice quality as most VoIP devices only have small fixed jitter buffers sized to cope with jitter imposed on the local network connection.

The greatest benefit is obtained when the techniques are used in combination but elements two and three are optional.  If using the final element then the first element must be switch on since it is the awareness of conversation context within the Vocality unit which allows the de-jittering technique to take advantage by adding its small timestamp at the same time. By utilizing a small proportion of the significant bandwidth savings achieved by the RTP compression technique, the quality of the SIP call is preserved across bandwidth-sensitive networks which suffer from dramatic variation in packet-to-packet delays, thereby achieving  the ultimate goal of low bandwidth and high quality at the same time. By providing individual controls over each of these elements it is possible to optimize the bandwidth and quality of the transmission when it is implemented across a Vocality satellite network.