pjsua2 AudioMediaPlayer play wav file as call,but it no sound in call, how to resolve？

Question

the main environment is docker, python3.8, pjsua2

I use this code to get a call, in microsip can accept, but no sound

the main code is

#!/usr/bin/env python3

import sys
import os
import logging
import pjsua2 as pj

# ep = pj.Endpoint()
ep = None

class Call(pj.Call):
    def __init__(self, acc, call_id=pj.PJSUA_INVALID_ID):
        pj.Call.__init__(self, acc, call_id)
        self.call_state = None

    def onCallState(self, prm):
        ci = self.getInfo()
        print("==== ==== Call state: {}".format(ci.state))
        print("==== ==== Call stateText: {}".format(ci.stateText))
        # save call state to some class var
        self.call_state = ci.state

        # playing the wav file
        play_dev_med = pj.Endpoint.instance().audDevManager().getPlaybackDevMedia()
        print(f"==== ==== {play_dev_med}")
        player = pj.AudioMediaPlayer()
        player.createPlayer("/workspace/pjsua/input.16.wav")
        player.startTransmit(play_dev_med)


class Account(pj.Account):
    def __init__(self):
        super().__init__()


def main():
    # Create and initialize the library
    global ep
    ep_cfg = pj.EpConfig()


    ep_cfg.uaConfig.threadCnt = 0
    ep_cfg.uaConfig.mainThreadOnly = True

    ep_cfg.logConfig.level = 5
    ep_cfg.logConfig.consoleLevel = 5

    ep = pj.Endpoint()
    ep.libCreate()
    ep.libInit(ep_cfg)
    # ep.audDevManager().setNullDev()

    # Create SIP transport. Error handling sample is shown
    sipTpConfig = pj.TransportConfig()
    ep.transportCreate(pj.PJSIP_TRANSPORT_UDP, sipTpConfig)

    # Start the library
    ep.libStart()

    ep.audDevManager().setNullDev()

    acfg = pj.AccountConfig()
    acfg.idUri = "sip:[email protected]"
    cred = pj.AuthCredInfo("digest", "*", "505", 0, "505")
    acfg.sipConfig.authCreds.append(cred)

    # Create the account
    acc = Account()
    acc.create(acfg)

    ep.audDevManager().setNullDev()
    call = Call(acc)
    call_param = pj.CallOpParam()
    call_param.opt.audioCount = 1
    call_param.opt.videoCount = 0
    call_prm = pj.CallOpParam(True)
    call.makeCall("sip:[email protected]", call_prm)

    # while not call.done:
    while True:
        ep.libHandleEvents(10000)
        # check your saved call_state here
        if not call.call_state:
            continue
        if call.call_state == pj.PJSIP_INV_STATE_CONFIRMED:
            call.done = True

    # Destroy the library
    ep.libDestroy()


if __name__ == "__main__":
    main()

when oncallstate, the log show that

16:00:37.804            pjsua_aud.c  .....Creating file player: /workspace/pjsua/input.16.wav..

16:00:37.804           wav_player.c  ......File player '/workspace/pjsua/input.16.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=134KB

16:00:37.804            pjsua_aud.c  ......Player created, id=0, slot=1
16:00:37.804            pjsua_aud.c  .....Conf connect: 1 --> 0
16:00:37.804           conference.c  ......Port 1 (/workspace/pjsua/input.16.wav) transmitting to port 0 (Master/sound)

16:00:37.804            pjsua_aud.c  .....Destroying player 0..

I notice that the player is destroyed soon, and how to resolve it

when accept the call, it's no sound