SIP call processing method and system, computer equipment, medium

Abstract

The invention relates to an SIP call processing method and system, computer equipment and a medium, and the method comprises the steps that a voice gateway sends a call initiating message to a user-level switch after receiving the off-hook and called number dialing information of a calling terminal; when the voice gateway receives a response message sent by the user-level switch, the voice gatewayis started to play a local routing sound to the calling terminal; when the voice gateway receives the ringing message of the user-level switch, controlling to stop playing the local routing sound tothe calling terminal; and when an off-hook signal of the called terminal is detected, the user-level switch establishes a call connection with the voice gateway, and the calling terminal and the called terminal establish a call channel with the voice gateway through the user-level switch. After the user makes the call and before the ring-back tone playing message is received, the sound is played locally, so that the user knows that the call is in communication in time, the user can be effectively prevented from hanging up, and convenience is provided for the user to use.

Description

technical field [0001] The present application relates to the technical field of voice communication, in particular to a SIP call processing method and system, computer equipment, and a readable storage medium. Background technique [0002] Voice communication is a means of communication through voice and with the aid of transmission media. Voice communication systems of the prior art such as figure 1 As shown, the telephone is connected to the voice gateway through a POTS (analog telephone service) line, and the voice gateway is connected to an IP-PBX (IPPrivateBranchExchange IP user-level exchange) through a network, and SIP (Session Initiation Protocol) is used in the middle. [0003] The call flow for establishing a voice call is as follows: figure 2 As shown, the voice gateway user dials the called number, and after receiving the number, the voice gateway sends an INVITE message (invitation message) to the IP-PBX to initiate a call establishment request, and the IP-P...

AI Technical Summary

Problems solved by technology

[0004] In the prior art, since the IP-PBX (user-level exchange) processes the call information initiated by the voice gateway, it takes time to forward it to the called party, and the called party also needs time to process it after receiving the message. There will be a relatively long time interval between the IP-PBX 180ring message. Under normal circumstances, there will be a time interval of about 2s between 100trying and 180ring. In the case of a poor network, the time interval is longer; in the middle of this time interval, the voice gateway The calling user is always in the mute state, so the calling user cannot know whether the call is connected. Some users may mistakenly think that the call is not connected, so they hang up and dial again, resulting in poor user experience; it is inconvenient for users to use

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