Method and device for intelligent network self-adaption of SIP VOIP (Session Initiation Protocol VOIP) phone

Abstract

The invention provides an SIP VOIP phone intelligent network self-adaption method and device, and the method comprises the steps: a network monitoring module calls a network interface module to obtain a test message which is regularly transmitted to a communication opposite end by each network interface, and carries out the analysis to obtain a network statistical value, real-time audio and video quality data are obtained by analyzing a voice statistical value in an RTCP message or an RCPP-XR message received from an opposite terminal in the dialogue process; the adaptive module receives the network statistical value and the voice statistical value, calls an adaptive adjustment algorithm, and modifies corresponding video parameters and audio parameters; the intelligent network switching module receives the network statistical value and the voice statistical value and judges whether network switching and target switching are needed or not according to numerical values; the self-adaptive adjustment module and the intelligent network switching module call the SIP signaling module and send re-INVITE or UPDATE information to update IP port information of the RTP and new audio parameters to be used, and meanwhile, the intelligent network switching module calls the SIP signaling module to send a re-REGISTER command to perform re-registration with a new IP.

Description

Technical Field

[0001] The embodiments of the present invention generally relate to the field of VoIP phone technology, and more particularly to a SIP VoIP phone intelligent network adaptive method and apparatus. Background Technology

[0002] With the rapid development of internet communication technology, the SIP protocol has been widely used in VoIP telephone systems. However, existing SIP phones typically use preset fixed audio and video configurations, making it difficult to cope with dynamically changing network environments. When network conditions (such as bandwidth, latency, jitter, packet loss rate, etc.) deteriorate, existing phones often experience problems such as decreased call quality, audio distortion, video stuttering, or dropped calls. Although some SIP VoIP phones support multiple codecs and network protocol optimizations, the following problems still exist:

[0003] 1. Fixed parameters: When the network fluctuates, it cannot automatically adjust the audio and video quality, which can easily lead to dropped calls or a decrease in call quality.

[0004] 2. Lack of real-time network monitoring: Existing equipment lacks dynamic monitoring of key network parameters (such as bandwidth, latency, jitter, packet loss rate, etc.).

[0005] 3. Poor user experience: When the network environment changes frequently, users need to manually select the network interface and adjust the call parameters, which affects the smoothness and stability of the call. Summary of the Invention

[0006] To address the above issues, this invention solves the problem of lack of dynamic monitoring in existing technologies by detecting real-time network parameters; the intelligent algorithm automatically adjusts the audio and video parameters of the call according to the network status, so that the call can still maintain high quality when the network fluctuates; when the network signal is weak or congested, the system can automatically switch to a more stable network to prevent the call from being dropped.

[0007] According to an embodiment of the present invention, a method and apparatus for intelligent network adaptation of SIP VoIP phones are provided.

[0008] In a first aspect of the invention, a method for intelligent network adaptation of a SIP VoIP phone is provided. The method includes:

[0009] S01: The network monitoring module calls the network interface module to obtain the test messages that each network interface sends to the communication peer periodically, parses them to obtain network statistics, and obtains real-time audio and video quality data by analyzing the voice statistics in the RTCP or RTCP-XR messages received from the peer during the dialogue. The network statistics and voice statistics are then sent to the adaptive module and the intelligent network switching module.

[0010] S02: The adaptive module receives network statistics and voice statistics, calls the adaptive adjustment algorithm, and modifies the corresponding video and audio parameters.

[0011] S03: The intelligent network switching module receives network statistics and voice statistics, and determines whether to switch networks and the target network based on the values.

[0012] S04: The adaptive adjustment module and the intelligent network switching module call the SIP signaling module to send a re-INVITE or UPDATE message to update the RTP IP port information and the new audio parameters to be used. At the same time, the intelligent network switching module calls the SIP signaling module to send a re-REGISTER command to re-register with the new IP.

[0013] Furthermore, the network statistics mentioned in S01 include: packet loss rate, response message, and return time; the voice statistics include: bandwidth, latency, jitter, and packet loss rate.

[0014] Furthermore, the workflow of the adaptive adjustment algorithm described in S02 is as follows:

[0015] S021: Determine whether the received message type is RTCP or RTCP-XR;

[0016] S022: If it is an RTCP packet, extract the jitter value / latency value / packet loss rate value to determine whether video is used. If so, adjust the video and audio parameters according to the jitter value / latency value / packet loss rate; otherwise, adjust the audio parameters according to the jitter value / latency value / packet loss rate.

[0017] S023: If it is an RTCP-XR packet, determine whether it contains video quality indicators. If it does, adjust the video parameters according to the PSNR value and adjust the audio parameters according to the extracted MOS value. Otherwise, adjust the audio parameters according to the extracted MOS value.

[0018] Furthermore, the audio parameters include resolution and frame rate, and the video parameters include codec, bitrate, echo cancellation level, and noise suppression level.

[0019] In a second aspect of the invention, a device for intelligent network adaptation of a SIP VoIP phone is provided. The device includes:

[0020] Network monitoring module: It is used to call the network interface module to obtain the test messages that each network interface sends to the communication peer periodically, parse them to obtain network statistics, obtain real-time audio and video quality data by analyzing the voice statistics in the RTCP or RTCP-XR messages received from the peer during the dialogue, and send the network statistics and voice statistics to the adaptive module and the intelligent network switching module.

[0021] Adaptive module: Used to receive network statistics and voice statistics, call the adaptive adjustment algorithm, and modify the corresponding video and audio parameters;

[0022] Intelligent network switching module: It receives network statistics and voice statistics, and determines whether to switch networks and the target network based on the values.

[0023] SIP signaling module: When called by the adaptive adjustment module and the intelligent network switching module, it sends re-INVITE or UPDATE messages to update the RTP IP port information and the new audio parameters to be used. At the same time, the intelligent network switching module calls the SIP signaling module to send the re-REGISTER command to re-register with the new IP.

[0024] Furthermore, the network statistics in the network monitoring module include: packet loss rate, response message, and return time; the voice statistics include: bandwidth, latency, jitter, and packet loss rate.

[0025] Furthermore, the workflow of the adaptive adjustment algorithm described in the adaptive module is as follows:

[0026] S021: Determine whether the received message type is RTCP or RTCP-XR;

[0027] S022: If it is an RTCP packet, extract the jitter value / latency value / packet loss rate value to determine whether video is used. If so, adjust the video and audio parameters according to the jitter value / latency value / packet loss rate; otherwise, adjust the audio parameters according to the jitter value / latency value / packet loss rate.

[0028] S023: If it is an RTCP-XR packet, determine whether it contains video quality indicators. If it does, adjust the video parameters according to the PSNR value and adjust the audio parameters according to the extracted MOS value. Otherwise, adjust the audio parameters according to the extracted MOS value.

[0029] Furthermore, the audio parameters include resolution and frame rate, and the video parameters include codec, bitrate, echo cancellation level, and noise suppression level.

[0030] The above-mentioned English abbreviations are explained as follows:

[0031] SIP: Session Initiation Protocol

[0032] VoIP: Voice Over Internet Protocol

[0033] RTCP: Real-time Transport Control Protocol

[0034] RTCP-XR: Real-time Transport Control Protocol-Extended Reports

[0035] re-INVITE: A message requesting a re-invite.

[0036] UPDATE: Media Parameter Update Request Message

[0037] RTP: Real-time Transport Protocol

[0038] re-REGISTER: Re-registration request message

[0039] This invention solves the problem of lack of dynamic monitoring in the prior art by detecting real-time network parameters; the intelligent algorithm automatically adjusts the audio and video parameters of the call according to the network status, so that the call can still maintain high quality when the network fluctuates; when the network signal is weak or congested, the system can automatically switch to a more stable network to prevent the call from being dropped.

[0040] It should be understood that the description in the Summary of the Invention is not intended to limit the key or essential features of the embodiments of the present invention, nor is it intended to restrict the scope of the invention. Other features of the invention will become readily apparent from the following description. Attached Figure Description

[0041] The above and other features, advantages, and aspects of the various embodiments of the present invention will become more apparent from the accompanying drawings and the following detailed description. Wherein:

[0042] Figure 1 A flowchart of a method for intelligent network adaptation of SIP VoIP phones according to an embodiment of the present invention is shown;

[0043] Figure 2 A system architecture diagram according to an embodiment of the present invention is shown;

[0044] Figure 3 A schematic diagram of a network monitoring module according to an embodiment of the present invention is shown;

[0045] Figure 4 A flowchart of the adaptive algorithm according to an embodiment of the present invention is shown;

[0046] Figure 5 A block diagram of a device for intelligent network adaptation of SIP VoIP phones according to an embodiment of the present invention is shown. Detailed Implementation

[0047] To make the objectives, technical solutions, and advantages of the embodiments of the present invention clearer, the technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0048] According to embodiments of the present invention, a method and apparatus for intelligent network adaptation of SIP VoIP phones are proposed. By detecting real-time network parameters, the lack of dynamic monitoring in the prior art is solved. The intelligent algorithm automatically adjusts the audio and video parameters of the call according to the network status, so that the call can still maintain high quality when the network fluctuates. When the network signal is weak or congested, the system can automatically switch to a more stable network to prevent the call from being dropped.

[0049] The principles and spirit of the present invention will be explained in detail below with reference to several representative embodiments.

[0050] Figure 1 This is a schematic flowchart of a SIP VoIP phone intelligent network adaptive method according to an embodiment of the present invention. The method includes:

[0051] S01: The network monitoring module calls the network interface module to obtain the test messages that each network interface sends to the communication peer periodically, parses them to obtain network statistics, and obtains real-time audio and video quality data by analyzing the voice statistics in the RTCP or RTCP-XR messages received from the peer during the dialogue. The network statistics and voice statistics are then sent to the adaptive module and the intelligent network switching module.

[0052] S02: The adaptive module receives network statistics and voice statistics, calls the adaptive adjustment algorithm, and modifies the corresponding video and audio parameters.

[0053] S03: The intelligent network switching module receives network statistics and voice statistics, and determines whether to switch networks and the target network based on the values.

[0054] S04: The adaptive adjustment module and the intelligent network switching module call the SIP signaling module to send a re-INVITE or UPDATE message to update the RTP IP port information and the new audio parameters to be used. At the same time, the intelligent network switching module calls the SIP signaling module to send a re-REGISTER command to re-register with the new IP.

[0055] It should be noted that although the operation of the method of the present invention has been described in a specific order in the above embodiments and figures, this does not require or imply that the operations must be performed in that specific order, or that all the operations shown must be performed to achieve the desired result. Additionally or alternatively, certain steps may be omitted, multiple steps may be combined into one step, and / or one step may be broken down into multiple steps.

[0056] To provide a clearer explanation of the above-described SIP VoIP phone intelligent network adaptive method, a specific embodiment will be used for illustration below. However, it is worth noting that this embodiment is only for better illustrating the present invention and does not constitute an improper limitation of the present invention.

[0057] The following specific example will further illustrate the method of intelligent network adaptation for SIP VoIP phones:

[0058] System architecture such as Figure 2 As shown, this illustrates the interactions and workflows between the modules.

[0059] Environment description:

[0060] Hardware requirements:

[0061] 1. Processor and memory: Since the adaptive adjustment module and related processing do not require much processor power, it is recommended to configure a CPU with more than 2 cores and at least 128MB of memory.

[0062] 2. Audio Module: Includes a built-in speaker, microphone, and headphone jack for supporting voice calls. The audio module should have echo cancellation and noise suppression functions to ensure call quality.

[0063] 3. Network interface: Supports at least two of the following: wired and wireless network connections and IoT connections (supporting 4G / 5G networks).

[0064] Software requirements:

[0065] 1. Operating System: The device runs an embedded operating system, such as Linux or other real-time operating systems, to provide a stable system operating environment.

[0066] 2. SIP Protocol Stack: Integrates a complete SIP protocol stack, including client and server functions, and supports the standard SIPRFC specification and common VoIP codecs, such as G.711 and G.729.

[0067] 3. User interface: HTML5-based web management interface or local graphical interface to simplify device configuration and management.

[0068] Functionality and performance were verified using an existing SIP phone terminal device without hardware modification. The configuration of this SIP phone terminal system is as follows:

[0069] CPU: Quad-core Cortex-A35 @ 1.3GHz

[0070] Memory: 256MB DDR3

[0071] Storage: 256MB Nand Flash

[0072] Network: Gigabit Ethernet port / Wi-Fi adapter / USB interface IoT 4G adapter

[0073] Operating System: Embedded Linux

[0074] Software modifications:

[0075] 1. Add a network monitoring module to handle this.

[0076] 2. Add hook processing to the RTCP / RTCP-XR receive module

[0077] 3. Create threads to monitor network quality across multiple interfaces in real time.

[0078] 4. Implement the adaptive adjustment module algorithm

[0079] 5. Added network switching module

[0080] The user is making a video call using a SIP VoIP phone connected to multiple network interfaces (such as wired, Wi-Fi, 4G / 5G). During the call, if network conditions are unstable, the SIP phone will dynamically adjust the audio and video encoding parameters and network interface based on RTCP / RTCP-XR data feedback and the quality of different network interfaces to ensure stable call quality.

[0081] When a user's SIP phone starts up, it first automatically detects the available network interfaces, including Ethernet, Wi-Fi, and 4G / 5G networks. The system prioritizes the network interface with the highest bandwidth and lowest latency.

[0082] At the start of a SIP session, the system initializes the audio codec to G.722, the video codec to H.264, the video resolution to 1080p, and the frame rate to 30fps, based on a preliminary assessment of network conditions.

[0083] like Figure 3 The diagram shows a schematic of the network monitoring module. The network monitoring module calls the network interface module to obtain the test packets periodically sent by each network interface to the communication peer, parses them to obtain network statistics, and obtains real-time audio and video quality data by analyzing the voice statistics in the RTCP or RTCP-XR packets received from the peer during the dialogue. The network statistics and voice statistics are then sent to the adaptive module and the intelligent network switching module. The content of the RTCP packets is shown in Table 1, and the content of the RTCP-XR packets is shown in Table 2.

[0084] Table 1

[0085]

[0086]

[0087] Table 2

[0088]

[0089] If RTCP / RTCP-XR reports increased network jitter or a packet loss rate exceeding a preset threshold (e.g., above 5%), the system detects a deterioration in network conditions.

[0090] The adaptive module receives network and voice statistics, invokes the adaptive adjustment algorithm, and modifies the corresponding video and audio parameters. Specifically, the levels of each indicator are determined according to Table 3:

[0091] Table 3

[0092] Ideal (Excellent) Acceptable (Good) Note (in Chinese) Not recommended (poor) Jitter value 0~30ms 30-50ms 50~100ms More than 100ms Latency <20ms 20~100ms 100~150ms More than 150ms Packet loss rate (PLR) 0％ 0～1％ 1～3％ More than 3% Audio Quality (MOS) 4.0～5.0 3.5～4.0 3.0～3.5 <3.0 Video quality (PSNR) >40dB 30~40dB 20-30dB <20dB

[0093] The MOS (Mean Opinion Score) and PSNR (Peak Signal-to-Noise Ratio) values ​​are obtained from RTCP-XR. If RTCP-XR is enabled, these two metrics can be used to determine audio and video parameter adjustments; otherwise, the Jitter / Latency / PLR values ​​obtained from RTCP are used.

[0094] Table 4 shows the rules for adjusting video parameters according to PSNR:

[0095] Table 4

[0096] PSNR level Video resolution Frame rate (FPS) Other parameter adjustments excellent 1080p or higher 30～60 High-quality echo cancellation, low noise suppression good 720p 30 Moderate noise suppression, standard echo cancellation middle 480p 20～30 Strong noise suppression, dynamic echo cancellation Difference 360p 15～20 Enhanced noise suppression and dynamic frame rate adjustment

[0097] Table 5 shows the rules for adjusting audio parameters according to MOS:

[0098] Table 5

[0099] MOS level audio codec Other parameter adjustments excellent G.722, Opus (High Quality) High-quality echo cancellation, low noise suppression good G.711, G.722, Opus (Medium Quality) Moderate noise suppression, standard echo cancellation middle G.729, Opus (low bitrate mode) Strong noise suppression, dynamic echo cancellation Difference G.729,iLBC Enhanced noise suppression and dynamic frame rate adjustment

[0100] Table 6 shows the video parameters to be adjusted according to Jitter / Latency / PLR.

[0101] Table 6

[0102] Jitter / Latency / PLR Levels Video resolution Frame rate (FPS) Other parameter adjustments excellent 1080p or higher 30～60 High-quality echo cancellation, low noise suppression good 720p 30 Moderate noise suppression, standard echo cancellation middle 480p 20～30 Strong noise suppression, dynamic echo cancellation Difference 360p 15～20 Enhanced noise suppression and dynamic frame rate adjustment

[0103] Table 7 shows how to adjust audio parameters according to Jitter / Latency / PLR:

[0104] Table 7

[0105]

[0106]

[0107] like Figure 4 As shown, the workflow of the adaptive adjustment algorithm is as follows:

[0108] Determine whether the received message type is RTCP or RTCP-XR;

[0109] If it is an RTCP packet, extract the jitter value / latency value / packet loss rate value to determine whether video is used. If so, adjust the video and audio parameters according to the jitter value / latency value / packet loss rate; otherwise, adjust the audio parameters according to the jitter value / latency value / packet loss rate.

[0110] If it is an RTCP-XR packet, determine whether it contains video quality metrics. If it does, adjust the video parameters according to the PSNR value and adjust the audio parameters according to the extracted MOS value; otherwise, adjust the audio parameters according to the extracted MOS value.

[0111] Specifically, audio adjustments: When RTCP data shows an increase in packet loss rate, the phone automatically switches from G.722 to the more packet-loss-resistant G.729 codec. Simultaneously, the audio frame length is slightly increased, and the transmission frequency is reduced to alleviate network load.

[0112] Video Adjustment: The video resolution is automatically reduced to 720p, and the frame rate is reduced from 30fps to 20fps to reduce data transmission and ensure smooth video playback.

[0113] QoS optimization: When the system detects an increase in packet loss, it automatically adjusts QoS parameters and marks the priority of RTP packets to ensure that voice streams are transmitted first.

[0114] The intelligent network switching module receives network statistics and voice statistics, and determines whether to switch networks and the target network based on the values.

[0115] Specifically, when the packet loss rate reported by RTCP / RTCP-XR continues to rise and the latency exceeds 150ms, the system considers the current network interface to be of poor quality and begins to detect the real-time status of other available interfaces.

[0116] The system detected that the Wi-Fi network signal strength was more stable than the current wired network, with better bandwidth and latency, and decided to switch to the Wi-Fi interface.

[0117] The handover process uses the re-INVITE mechanism in SIP to update the IP address and port of the RTP stream to the Wi-Fi network in the background without interrupting the call.

[0118] After the switch was completed, the user's call remained stable, the audio and video quality was restored to its best state, the video resolution was adjusted back to 1080p, and the frame rate was restored to 30fps.

[0119] The adaptive adjustment module and the intelligent network switching module call the SIP signaling module to send re-INVITE or UPDATE messages to update the RTP IP port information and the new audio parameters to be used. At the same time, the intelligent network switching module calls the SIP signaling module to send a re-REGISTER command to re-register with the new IP.

[0120] The system continuously monitors network conditions via RTCP / RTCP-XR, dynamically adjusting audio and video parameters to ensure call quality. Whenever network conditions deteriorate or the network quality of different interfaces changes, the system reassesses and makes corresponding adjustments.

[0121] If all network interfaces are unstable, the system will further reduce the quality of the codecs, such as reducing the video resolution to 480p, or even turning off the video and keeping only the audio.

[0122] Specifically, it can control the RTCP or RTCP-XR packets from the phone to be sent to a dedicated server for media quality analysis (or integrated as a service into a media gateway or SIP server). Based on the media quality analysis, the server sends UPDATE messages to both parties in the call to automatically adjust the audio and video parameters.

[0123] Based on the same inventive concept, this invention also proposes a device for intelligent network adaptation of SIP / VoIP phones. The implementation of this device can be found in the implementation of the method described above; repeated details will not be repeated. Figure 2 As shown, the device 100 includes:

[0124] Network monitoring module 101: It is used to call the network interface module to obtain the test messages that each network interface sends to the communication peer periodically, parse them to obtain network statistics, obtain real-time audio and video quality data by analyzing the voice statistics in the RTCP or RTCP-XR messages received from the peer during the dialogue, and send the network statistics and voice statistics to the adaptive module and the intelligent network switching module.

[0125] Adaptive module 102: Used to receive network statistics and voice statistics, call the adaptive adjustment algorithm, and modify the corresponding video and audio parameters;

[0126] Intelligent network switching module 103: Used to receive network statistics and voice statistics, and determine whether to switch networks and the target network based on the values;

[0127] SIP signaling module 104: When called by the adaptive adjustment module and the intelligent network switching module, it sends re-INVITE or UPDATE messages to update the RTP IP port information and the new audio parameters to be used. At the same time, the intelligent network switching module calls the SIP signaling module to send the re-REGISTER command to re-register with the new IP.

[0128] This invention proposes a device for intelligent network adaptation of SIP VoIP phones. By detecting real-time network parameters, it solves the problem of lack of dynamic monitoring in the prior art. The intelligent algorithm automatically adjusts the audio and video parameters of the call according to the network status, so that the call can still maintain high quality when the network fluctuates. When the network signal is weak or congested, the system can automatically switch to a more stable network to prevent the call from being dropped.

[0129] While the spirit and principles of the invention have been described with reference to several specific embodiments, it should be understood that the invention is not limited to the disclosed specific embodiments, and the division of aspects does not imply that features in these aspects cannot be combined for benefit; such division is merely for ease of description. The invention is intended to cover various modifications and equivalent arrangements included within the spirit and scope of the appended claims.

[0130] Regarding the limitation of the scope of protection of this invention, those skilled in the art should understand that various modifications or variations that can be made by those skilled in the art without creative effort based on the technical solution of this invention are still within the scope of protection of this invention.

Claims

1. A method for intelligent network adaptive operation of SIP / VoIP phones, characterized in that, The method includes: S01: The network monitoring module calls the network interface module to obtain the test messages that each network interface sends to the communication peer periodically, parses them to obtain network statistics, and obtains real-time audio and video quality data by analyzing the voice statistics in the RTCP or RTCP-XR messages received from the peer during the dialogue. The network statistics and voice statistics are then sent to the adaptive module and the intelligent network switching module. S02: The adaptive module receives network statistics and voice statistics, calls the adaptive adjustment algorithm, and modifies the corresponding video and audio parameters. S03: The intelligent network switching module receives network statistics and voice statistics, and determines whether to switch networks and the target network based on the values. S04: The adaptive adjustment module and the intelligent network switching module call the SIP signaling module to send a re-INVITE or UPDATE message to update the RTP IP port information and the new audio parameters to be used. At the same time, the intelligent network switching module calls the SIP signaling module to send a re-REGISTER command to re-register with the new IP.

2. The method for intelligent network adaptation of SIP VoIP phones according to claim 1, characterized in that, The network statistics mentioned in S01 include: packet loss rate, response message, and return time. The voice statistics include: bandwidth, latency, jitter, and packet loss rate.

3. The method for intelligent network adaptation of SIP VoIP phones according to claim 1, characterized in that, The workflow of the adaptive adjustment algorithm described in S02 is as follows: S021: Determine whether the received message type is RTCP or RTCP-XR; S022: If it is an RTCP packet, extract the jitter value / latency value / packet loss rate value to determine whether video is used. If so, adjust the video and audio parameters according to the jitter value / latency value / packet loss rate; otherwise, adjust the audio parameters according to the jitter value / latency value / packet loss rate. S023: If it is an RTCP-XR packet, determine whether it contains video quality indicators. If it does, adjust the video parameters according to the PSNR value and adjust the audio parameters according to the extracted MOS value. Otherwise, adjust the audio parameters according to the extracted MOS value.

4. The method for intelligent network adaptation of a SIP VoIP phone according to claim 3, characterized in that, The audio parameters include resolution and frame rate, and the video parameters include codec, bitrate, echo cancellation level, and noise suppression level.

5. A device for intelligent network adaptive operation of a SIP / VoIP phone, characterized in that, The device includes: Network monitoring module: It is used to call the network interface module to obtain the test messages that each network interface sends to the communication peer periodically, parse them to obtain network statistics, obtain real-time audio and video quality data by analyzing the voice statistics in the RTCP or RTCP-XR messages received from the peer during the dialogue, and send the network statistics and voice statistics to the adaptive module and the intelligent network switching module. Adaptive module: Used to receive network statistics and voice statistics, call the adaptive adjustment algorithm, and modify the corresponding video and audio parameters; Intelligent network switching module: It receives network statistics and voice statistics, and determines whether to switch networks and the target network based on the values. SIP signaling module: When called by the adaptive adjustment module and the intelligent network switching module, it sends re-INVITE or UPDATE messages to update the RTP IP port information and the new audio parameters to be used. At the same time, the intelligent network switching module calls the SIP signaling module to send the re-REGISTER command to re-register with the new IP.

6. The device for intelligent network adaptation of a SIP / VoIP phone according to claim 5, characterized in that, The network statistics in the network monitoring module include: packet loss rate, response messages, and return time. The voice statistics include: bandwidth, latency, jitter, and packet loss rate.

7. The device for intelligent network adaptation of SIP VoIP phones according to claim 5, characterized in that, The workflow of the adaptive adjustment algorithm described in the adaptive module is as follows: S021: Determine whether the received message type is RTCP or RTCP-XR; S022: If it is an RTCP packet, extract the jitter value / latency value / packet loss rate value to determine whether video is used. If so, adjust the video and audio parameters according to the jitter value / latency value / packet loss rate; otherwise, adjust the audio parameters according to the jitter value / latency value / packet loss rate. S023: If it is an RTCP-XR packet, determine whether it contains video quality indicators. If it does, adjust the video parameters according to the PSNR value and adjust the audio parameters according to the extracted MOS value. Otherwise, adjust the audio parameters according to the extracted MOS value.

8. The device for intelligent network adaptation of a SIP / VoIP phone according to claim 7, characterized in that, The audio parameters include resolution and frame rate, and the video parameters include codec, bitrate, echo cancellation level, and noise suppression level.

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