Voice Gateway Packet Combining for VoIP Header Overhead Reduction
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Solution Overview
Problem
Current methods for compressing VoIP headers in packet-switched networks, such as cRTP, EcRTP, and ROHC, suffer from issues like call interruption, congestion collapse, and inability to handle packet loss or misordering, especially in high latency networks like satellite communications, and are not compatible with both IPv4 and IPv6 networks simultaneously.
Innovation Solution
Combining voice media payloads into a single packet stream with a single set of IP headers, which are transmitted once and stored at voice gateways, reducing the need for retransmission and minimizing header overhead, while using voice trunks to manage bandwidth and prioritize calls using MLPP and CAC mechanisms.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Reliability
If individual packetization is used for each VoIP call, then call reliability is maintained, but bandwidth efficiency deteriorates due to substantial IP/UDP/RTP header overheads
Solution Approach 1:
The patent combines multiple VoIP call packets into a single packet stream, merging the IP headers and UDP/RTP headers so that instead of each call carrying its own full header overhead, all calls share a single set of headers. This merging approach maintains call reliability through proper identification mechanisms while dramatically improving bandwidth efficiency by eliminating redundant header data.
2Loss of energy
If header compression is applied using cRTP, EcRTP, or ROHC, then bandwidth efficiency is improved, but call reliability deteriorates due to context loss and congestion collapse
Solution Approach 1:
The patent performs preliminary identification and registration of each call's IP header and UDP/RTP header data before compression. By预先 identifying and storing the unique characteristics of each call's headers, the system can reliably reconstruct and decompress headers even in high-latency networks, preventing context loss and congestion collapse while maintaining bandwidth efficiency.
3Reliability
If full headers are retransmitted to refresh context at the decompressor, then context loss is recovered, but network congestion worsens due to additional traffic
Solution Approach 1:
The patent extracts and identifies the unique IP header and UDP/RTP header data for each individual call before compression. By separating and pre-identifying these call-specific header elements, the system eliminates the need for full header retransmission to refresh context, as the decompressor can reliably reconstruct headers from the extracted identification data, thereby avoiding additional network traffic and congestion.
4Adaptability or versatility
If cRTP or EcRTP is used for header compression, then IPv4 and IPv6 can be used simultaneously, but processor intensity increases causing capacity issues
Solution Approach 1:
The patent performs preliminary identification and registration of IP header characteristics for each call before compression occurs. By预先 identifying and storing the unique IP header data (whether IPv4 or IPv6), the system reduces the computational complexity during actual compression and decompression operations, as the identification data is already available and can be directly used without intensive real-time processing, thereby reducing processor intensity while maintaining multi-protocol compatibility.
Data Source
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AI summary
The invention relates to a voice gateway (10) for transmitting voice data to a receiving voice gateway (12) in a packet-switched network (14) by establishing a voice trunk (16) comprising: means (26) for receiving packets transmitting voice data between the two voice gateways, wherein each received packet comprises header data and voice media payload, means (28) for establishing a voice trunk between the two voice gateways based on the header data of the received packets, means (30) for extracting and combining the voice media payloads of the received packets into a single packet, wherein the single packet comprises for a voice call a dedicated channel data of the established voice trunk comprising an identifier of the respective voice call and the voice media payload of the respective voice call, means (32) for adding a single set of header data to the single packet, wherein the single set of header data comprises base information for synchronizing the flow of data between the two voice gateways, and means (34) for transmitting the single packet to the receiving voice gateway via the voice trunk.