IMS Voice-Packet Switching for Calls in Poor Wireless Conditions
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Solution Overview
Problem
In poor wireless communications environments, real-time voice calls experience high packet loss rates and delays due to low channel bandwidth, which are not adequately addressed by existing encoding methods.
Innovation Solution
A data transmission method that switches between audio streams and voice message packets based on signal strength, using an access network device to manage channel resources and enable call-to-message functions to reduce packet loss and delay.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If real-time call encoding scheme is used to encode audio stream, then call voice quality is improved, but packet loss rate increases and delay occurs when channel bandwidth is low
Solution Approach 1:
The patent implements dynamic switching between real-time call encoding scheme and extremely low bit rate encoding scheme based on channel bandwidth conditions. The access network device determines the signal strength and dynamically adjusts the encoding scheme, transitioning from static to dynamic operation to adapt to changing network conditions and resolve the contradiction between voice quality and packet loss rate
Solution Approach 2:
The patent changes the encoding parameters (bit rate) based on channel conditions. When channel bandwidth is sufficient, it uses real-time call encoding with higher bit rate for better quality; when bandwidth is limited, it switches to extremely low bit rate encoding to reduce data amount and prevent packet loss, thus resolving the contradiction through parameter adaptation
2Reliability
If extremely low bit rate encoding scheme is used to reduce data amount, then packet loss rate decreases, but call voice quality deteriorates
Solution Approach 1:
The system dynamically switches between encoding schemes based on real-time channel conditions. When bandwidth is sufficient, it uses high-quality real-time encoding; when bandwidth is limited, it switches to low-bit-rate encoding. This dynamic adaptation resolves the contradiction by selecting the appropriate encoding quality based on network capacity
Solution Approach 2:
The access network device monitors channel bandwidth and signal strength continuously, using this feedback to determine which encoding scheme to apply. The feedback mechanism allows the system to adjust encoding parameters in real-time, ensuring optimal balance between voice quality and packet loss prevention under different network conditions
3Productivity
If audio stream is transmitted through low bandwidth channel, then data transmission capacity is充分利用, but delay increases
Solution Approach 1:
The patent changes the data transmission parameters by switching between audio stream format and voice message packet format based on channel bandwidth. When bandwidth is limited, it transforms the transmission format to voice message packets with smaller data amounts, reducing both the data volume and transmission delay while adapting to the available bandwidth
Data Source
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AI summary
Embodiments of this application provide a data transmission method and a related device, to implement a call between a first call end and a second call end when a communications environment is poor. The method in embodiments of this application includes: An access network device establishes a call between a first call end and a second call end by using an IMS network. The access network device receives a first voice message packet from the first call end. The first voice message packet is sent after the first call end determines that a first signal strength is less than a first threshold, and the first signal strength is a signal strength of the access network device at the first call end. The access network device transmits the first voice message packet to a media processing network element, to implement the call between the first call end and the second call end by using the media processing network element.