voice-gateway/voip/hello.py

#!/usr/bin/env python3
# encoding:utf-8

import time
import json
import wave
import traceback
import pjsua2 as pj

calls = {}
recording_file = '/code/pjsip-apps/src/swig/python/incoming_call.wav'
player_file = '/code/pjsip-apps/src/swig/python/test111.wav'


# Subclass to extend the Account and get notifications etc.
class Account(pj.Account):
    def __init__(self):
        pj.Account.__init__(self)

    def onRegState(self, prm):
        print("***OnRegState: " + prm.reason)

    def onIncomingCall(self, prm):
        print("***onIncomingCall: ", prm.callId)

        call = MyCall(self, prm.callId)
        call_op_param = pj.CallOpParam(True)
        call_op_param.statusCode = pj.PJSIP_SC_OK
        call.answer(call_op_param)
        calls[prm.callId] = call


class MyAudioMediaPort(pj.AudioMediaPort):
    def __init__(self):
        pj.AudioMediaPort.__init__(self)
        # 打开一个 .pcm 文件来保存音频流（可选：保存为 .wav）
        self.wav = wave.open(f"{recording_file}", "wb")
        self.wav.setnchannels(1)
        self.wav.setsampwidth(2)  # 假设每个样本是 16 位（2 字节）
        self.wav.setframerate(16000)

    def onFrameRequested(self, frame):
        print("Request audio frame:", frame)

    def onFrameReceived(self, frame):
        self.wav.writeframes(bytes(frame.buf))
        # print("Received audio frame:", frame.buf, frame.size)


class MyCall(pj.Call):

    def __init__(self, acc, call_id):
        pj.Call.__init__(self, acc, call_id)
        self.call_id = call_id
        self.audio_port = None
        self.aud_med = None
        self.recorder = None
        self.player = None  # 用于播放录音的媒体播放器

    def onDtmfDigit(self, prm):
        digit = prm.digit
        print(f"Received DTMF digit: {digit}")

    def onCallState(self, prm):
        call_info = self.getInfo()
        print("Call state: ", call_info.stateText, dir(call_info))
        # if call_info.state == pj.PJSIP_INV_STATE_CONFIRMED:
        #     # 当呼叫状态为已确认（即接通）
        #     audio_media = self.getAudioMedia()  # 获取音频媒体对象
        #     audio_media.startTransmit(audio_media)  # 这里可以对音频流进行处理
        #     print ('11111111111')
        if call_info.state == pj.PJSIP_INV_STATE_DISCONNECTED:
            print("通话结束")
            self.delete()

    def onCallMediaState(self, prm):
        call_info = self.getInfo()
        print("Call Media state: ", call_info.stateText)
        for media in call_info.media:
            if media.type == pj.PJMEDIA_TYPE_AUDIO and \
               (media.status == pj.PJSUA_CALL_MEDIA_ACTIVE):
                self.aud_med = self.getAudioMedia(media.index)
                try:
                    self.audio_port = MyAudioMediaPort()
                    self.audio_port.createPort("Incoming Call Port", self.build_audio_format())
                    self.aud_med.startTransmit(self.audio_port)

                    # 录制来电声音
                    # self.recorder = pj.AudioMediaRecorder()
                    # self.recorder.createRecorder(recording_file)
                    # self.aud_med.startTransmit(self.recorder)
                    # self.send_audio_to_asr()

                    # 播放其它录音文件
                    self.player = pj.AudioMediaPlayer()
                    self.player.createPlayer(player_file)
                    self.player.startTransmit(self.aud_med)
                    # 显示播放进度
                    # self.display_playback_progress()
                except Exception as e:
                    traceback.print_exc()

    def build_audio_format(self):
        fmt = pj.MediaFormatAudio()
        fmt.type = pj.PJMEDIA_TYPE_AUDIO
        fmt.id = pj.PJMEDIA_FORMAT_PCM
        fmt.clockRate = 16000  # 采样率
        fmt.channelCount = 1  # 通道数
        fmt.frameTimeUsec = 20000  # 每帧的时间（20 毫秒）
        fmt.bitsPerSample = 16  # 每个采样的位数
        return fmt

    def display_playback_progress(self):
        while self.player:
            # 获取当前播放位置和总时长
            current_pos = self.player.getPos()
            player_info = self.player.getInfo()
            print (current_pos, player_info.sizeBytes, player_info.sizeSamples)
            total_duration = player_info.sizeBytes

            if total_duration > 0:
                progress = (current_pos / total_duration) * 100
                print(f"播放进度: {progress:.2f}%")

            time.sleep(1)  # 每隔1秒更新一次进度

    def stopMedia(self):
        pass
        # if self.player:
        #     self.player.stopTransmit(self.aud_med)
        #     self.player = None
        # if self.recorder:
        #     self.recorder.stopTransmit(self.aud_med)
        #     self.recorder = None

    def delete(self):
        self.stopMedia()


# pjsua2 test function
def pjsua2_test():
    # Create and initialize the library
    ep = pj.Endpoint()
    ep_cfg = pj.EpConfig()
    try:
        ep.libCreate()
        ep.libInit(ep_cfg)

        aud_dev_mgr = ep.audDevManager()
        aud_dev_mgr.setNullDev()  # 使用虚拟音频设备（如果没有实际设备）


        # Create SIP transport. Error handling sample is shown
        sipTpConfig = pj.TransportConfig()
        sipTpConfig.port = 30506
        ep.transportCreate(pj.PJSIP_TRANSPORT_UDP, sipTpConfig)
        # Start the library
        ep.libStart()

        acfg = pj.AccountConfig()
        acfg.idUri = "sip:1001@172.16.12.16:5060"
        acfg.regConfig.registrarUri = "sip:172.16.12.16:5060"
        cred = pj.AuthCredInfo("digest", "*", "1001", 0, "slibra@#123456")
        # acfg.idUri = "sip:1013@test-bot.shuiditech.com:30506"
        # acfg.regConfig.registrarUri = "sip:test-bot.shuiditech.com:30506"
        # cred = pj.AuthCredInfo("digest", "*", "1013", 0, "000000")
        acfg.sipConfig.authCreds.append(cred)
        # Create the account
        acc = Account()
        acc.create(acfg)

        try:
            while True:
                ep.libHandleEvents(100)
        except KeyboardInterrupt:
            print('Existing...')
        # while True:
        #     # Here we don't have anything else to do..
        #     time.sleep(10);
    finally:
        # Destroy the library
        ep.libDestroy()


#
# main()
#
if __name__ == "__main__":
    pjsua2_test()