我使用 PJSUA2 与应用程序建立成熟的 SIP 呼叫，该应用程序应该：
即使呼叫已连接到服务器（我可以看到呼叫、其 ID、一些日志等），但由于某种原因无法建立媒体会话。
这是我运行的代码（python 3.11）
import pjsua2 as pj
import time
import random
endpoint_config = pj.EpConfig()
endpoint_config.logConfig.level = 9
endpoint_config.medConfig.sndClockRate = 8000
endpoint = pj.Endpoint()
endpoint.libCreate()
endpoint.libInit(endpoint_config)
print("Endpoint initialized..")
# Create SIP transport
sip_transport_config = pj.TransportConfig()
transport_type = pj.PJSIP_TRANSPORT_TCP
sip_transport_config.port = 5060
endpoint.transportCreate(transport_type, sip_transport_config)
print("Transport created..")
# Start the library
endpoint.libStart()
print("Endpoint started..")
endpoint.libHandleEvents(10)
# Subclass to extend the Account and get notifications etc.
class MyAccount(pj.Account):
def __init__(self):
super().__init__()
def onRegState(self, prm):
ai = self.getInfo()
print("*** Register: code=" if ai.regIsActive else "*** Unregister: code=", prm.code)
class MyCall(pj.Call):
def __init__(self, account, call_id):
super().__init__(account, call_id)
self.acc = account
self.connected = False
def onCallState(self, prm):
time.sleep(5)
print(f"THE STATE OF THE CALL HAS CHANGED: {prm}")
call_info = self.getInfo()
self.connected = call_info.state == pj.PJSIP_INV_STATE_CONFIRMED
print("CONNECTED:", self.connected)
print("current state is " + str(call_info.state) + " " + call_info.stateText)
print("GET ID:", self.getId())
print("HAS MEDIA:", self.hasMedia())
def onMediaState(self):
print("ON MEDIA STATE")
# pjsua2 test function
def pj_test():
account_config.idUri = "sip:[email protected]"
account = MyAccount()
try:
account.create(account_config)
print("ACCOUNT?", account)
print("\nAccount created/registered..\n")
except pj.Error as err:
print("ERR acc:",err)
call = MyCall(account, pj.PJSUA_INVALID_ID)
prm = pj.CallOpParam(True)
try:
print("Attempting to make a call...")
# USE prm.txOption – Optional headers etc to be added to outgoing INVITE request.
call.makeCall("sip:[email protected]:5060;transport=tcp", prm) ## Dev
print("CALLINFO",callinfo := call.getInfo())
print("STATE:", callinfo.state)
print("CALLID:", callinfo.id)
print("GET ID:", call.getId())
print("HAS MEDIA:", call.hasMedia())
except Exception as err:
print("ERR:",err)
# call.dump()
endpoint.libDestroy()
if __name__ == "__main__":
pj_test()
这是日志：
11:13:56.697 os_core_unix.c !pjlib 2.13 for POSIX initialized
11:13:56.698 sip_endpoint.c .Creating endpoint instance...
11:13:56.699 pjlib .select() I/O Queue created (0x7f7eb4836ed8)
11:13:56.699 sip_endpoint.c .Module "mod-msg-print" registered
11:13:56.699 sip_transport.c .Transport manager created.
11:13:56.699 pjsua_core.c .PJSUA state changed: NULL --> CREATED
11:13:56.699 sip_endpoint.c .Module "mod-pjsua-log" registered
11:13:56.699 sip_endpoint.c .Module "mod-tsx-layer" registered
11:13:56.699 sip_endpoint.c .Module "mod-stateful-util" registered
11:13:56.699 sip_endpoint.c .Module "mod-ua" registered
11:13:56.699 sip_endpoint.c .Module "mod-100rel" registered
11:13:56.699 sip_endpoint.c .Module "mod-pjsua" registered
11:13:56.700 sip_endpoint.c .Module "mod-invite" registered
11:13:56.773 coreaudio_dev.c ..core audio detected 5 devices
11:13:56.777 coreaudio_dev.c .. dev_id 73: Soundcore Life Note (in=1, out=0) 16000Hz
11:13:56.777 coreaudio_dev.c .. dev_id 67: Soundcore Life Note (in=0, out=2) 44100Hz
11:13:56.785 coreaudio_dev.c .. dev_id 53: Built-in Microphone (in=2, out=0) 44100Hz
11:13:56.786 coreaudio_dev.c .. dev_id 43: Built-in Output (in=0, out=2) 44100Hz
11:13:56.795 coreaudio_dev.c .. dev_id 60: ZoomAudioDevice (in=2, out=2) 48000Hz
11:13:56.795 coreaudio_dev.c ..core audio initialized
11:13:56.796 pjlib ..select() I/O Queue created (0x7f7eb4841828)
11:13:56.801 sip_endpoint.c .Module "mod-evsub" registered
11:13:56.801 sip_endpoint.c .Module "mod-presence" registered
11:13:56.801 sip_endpoint.c .Module "mod-mwi" registered
11:13:56.801 sip_endpoint.c .Module "mod-refer" registered
11:13:56.801 sip_endpoint.c .Module "mod-pjsua-pres" registered
11:13:56.801 sip_endpoint.c .Module "mod-pjsua-im" registered
11:13:56.801 sip_endpoint.c .Module "mod-pjsua-options" registered
11:13:56.801 pjsua_core.c .1 SIP worker threads created
11:13:56.801 pjsua_core.c .pjsua version 2.13 for Darwin-21.6/x86_64 initialized
11:13:56.801 pjsua_core.c .PJSUA state changed: CREATED --> INIT
Endpoint initialized..
11:13:56.802 tcptp:5060 SIP TCP listener ready for incoming connections at 192.111.111.11:5060
Transport created..
11:13:56.802 pjsua_core.c PJSUA state changed: INIT --> STARTING
11:13:56.802 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
11:13:56.802 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
Endpoint started..
11:13:56.813 pjsua_acc.c Adding account: id=sip:[email protected]
11:13:56.813 pjsua_acc.c .Account sip:[email protected] added with id 0
ACCOUNT? <__main__.MyAccount; proxy of <Swig Object of type 'pj::Account *' at 0x10261a820> >
Account created/registered..
Attempting to make a call...
11:13:56.816 pjsua_call.c Making call with acc #0 to sip:[email protected]:5060;transport=tcp
11:13:56.816 pjsua_aud.c .Set sound device: capture=-1, playback=-2, mode=0, use_default_settings=0
11:13:56.816 pjsua_aud.c ..Opening sound device (speaker + mic) PCM@8000/1/20ms
11:13:56.833 coreaudio_dev.c ...Using VoiceProcessingIO audio unit
11:13:56.833 coreaudio_dev.c ...Warning: audio device id settings are ignored when using VPIO
11:13:58.514 coreaudio_dev.c ...core audio stream started
11:13:58.556 os_core_unix.c Info: possibly re-registering existing thread
11:13:58.677 tcpc0x7f7eb5856428 !.TCP client transport created
11:13:58.677 tcpc0x7f7eb5856428 .TCP transport 192.111.111.11:12121 is connecting to 12.115.110.139:5060...
11:13:58.677 pjsua_media.c .Call 0: initializing media..
11:13:58.702 pjsua_media.c ..RTCP socket bind() at 0.0.0.0:4001 error: Address already in use
11:13:58.702 pjsua_media.c ..RTP socket reachable at 192.111.111.11:4002
11:13:58.702 pjsua_media.c ..RTCP socket reachable at 192.111.111.11:4003
11:13:58.702 pjsua_media.c ..Media index 0 selected for audio call 0
11:13:58.702 udp0x7f7eb44500a0 ..UDP media transport created
11:13:58.703 pjsua_core.c ....TX 1432 bytes Request msg INVITE/cseq=5438 (tdta0x7f7eb48536a8) to TCP 12.115.110.139:5060:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.111.111.11:12121;rport;branch=z9hG4bKPj94DC87A8-F6E8-484A-8666-DA6FDF806591;alias
Max-Forwards: 70
From: sip:[email protected];tag=DF19B94B-5324-4EA6-93FC-E85CCB4E3083
To: sip:[email protected]
Contact: <sip:[email protected]:12121;transport=TCP;ob>
Call-ID: 53AF310B-070D-4F00-9AC9-5373F04A7C66
CSeq: 5438 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 777
v=0
o=- 3897447238 3897447238 IN IP4 192.111.111.11
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 96 97 98 99 3 0 8 9 100 101 120 121 122 123
c=IN IP4 192.111.111.11
b=TIAS:96000
a=rtcp:4003 IN IP4 192.111.111.11
a=sendrecv
a=rtpmap:96 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:98 speex/32000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:100 AMR/8000
a=rtpmap:101 opus/48000/2
a=fmtp:101 useinbandfec=1
a=rtpmap:120 telephone-event/16000
a=fmtp:120 0-16
a=rtpmap:121 telephone-event/8000
a=fmtp:121 0-16
a=rtpmap:122 telephone-event/32000
a=fmtp:122 0-16
a=rtpmap:123 telephone-event/48000
a=fmtp:123 0-16
a=ssrc:809471301 cname:51c99fb30c13cab8
--end msg--
11:13:58.810 tcpc0x7f7eb5856428 !TCP transport 192.111.111.11:12121 is connected to 12.115.110.139:5060
THE STATE OF THE CALL HAS CHANGED: <pjsua2.OnCallStateParam; proxy of <Swig Object of type 'pj::OnCallStateParam *' at 0x10261a610> >
CONNECTED: False
current state is 1 CALLING
GET ID: 0
HAS MEDIA: False
CALLINFO <pjsua2.CallInfo; proxy of <Swig Object of type 'pj::CallInfo *' at 0x10261a640> >
STATE: 1
CALLID: 0
GET ID: 0
HAS MEDIA: False
11:14:03.712 pjsua_core.c !Shutting down, flags=0...
11:14:03.712 pjsua_core.c PJSUA state changed: RUNNING --> CLOSING
11:14:03.712 pjsua_core.c .RX 393 bytes Response msg 100/INVITE/cseq=5438 (rdata0x7f7eb5856708) from TCP 12.115.110.139:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/TCP 192.111.111.11:12121;rport=12121;branch=z9hG4bKPj94DC87A8-F6E8-484A-8666-DA6FDF806591;alias;received=62.103.179.193
From: sip:[email protected];tag=DF19B94B-5324-4EA6-93FC-E85CCB4E3083
To: sip:[email protected]
Call-ID: 53AF310B-070D-4F00-9AC9-5373F04A7C66
CSeq: 5438 INVITE
Server: OmSBC
Content-Length: 0
--end msg--
11:14:03.712 pjsua_core.c .RX 1060 bytes Response msg 200/INVITE/cseq=5438 (rdata0x7f7eb5856708) from TCP 12.115.110.139:5060:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.111.111.11:12121;rport=12121;received=62.103.179.193;branch=z9hG4bKPj94DC87A8-F6E8-484A-8666-DA6FDF806591;alias
Record-Route: <sip:10.163.28.132:5080;transport=tcp;lr;r2=on;did=b31.63b2;nat=yes>
Record-Route: <sip:12.115.110.139;transport=tcp;lr;r2=on;did=b31.63b2;nat=yes>
Call-ID: 53AF310B-070D-4F00-9AC9-5373F04A7C66
From: <sip:[email protected]>;tag=DF19B94B-5324-4EA6-93FC-E85CCB4E3083
To: <sip:[email protected]>;tag=a1c3099f-9f54-48b9-affc-019adc7f44bc
CSeq: 5438 INVITE
Server: Asterisk PBX 18.7.1
Contact: <sip:10.163.29.13:5060;transport=TCP>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 195
v=0
o=- 3897447238 3897447240 IN IP4 63.35.69.252
s=Asterisk
c=IN IP4 63.35.69.252
t=0 0
m=audio 34112 RTP/AVP 0
a=maxptime:150
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtcp:34113
a=ptime:20
--end msg--
THE STATE OF THE CALL HAS CHANGED: <pjsua2.OnCallStateParam; proxy of <Swig Object of type 'pj::OnCallStateParam *' at 0x10261a580> >
CONNECTED: False
current state is 4 CONNECTING
GET ID: 0
HAS MEDIA: False
11:14:08.715 inv0x7f7eb5025ea8 ....SDP negotiation done: Success
11:14:08.715 pjsua_call.c .....Unable to create media session: Object is busy (PJ_EBUSY) [status=70011]
11:14:08.715 pjsua_core.c ........TX 552 bytes Request msg BYE/cseq=5439 (tdta0x7f7eb38562a8) to TCP 12.115.110.139:5060:
BYE sip:10.163.29.13:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 192.111.111.11:12121;rport;branch=z9hG4bKPj9F2572CC-3E19-4857-8D61-86F294CB0FA9;alias
Max-Forwards: 70
From: sip:[email protected];tag=DF19B94B-5324-4EA6-93FC-E85CCB4E3083
To: sip:[email protected];tag=a1c3099f-9f54-48b9-affc-019adc7f44bc
Call-ID: 53AF310B-070D-4F00-9AC9-5373F04A7C66
CSeq: 5439 BYE
Route: <sip:12.115.110.139;transport=tcp;lr;r2=on;did=b31.63b2;nat=yes>
Route: <sip:10.163.28.132:5080;transport=tcp;lr;r2=on;did=b31.63b2;nat=yes>
Content-Length: 0
从日志中可以看出：
Unable to create media session: Object is busy (PJ_EBUSY) [status=70011]我在这里看到了类似的答案，但这些答案适用于某些 Android 应用程序，这不是我正在研究的内容。解决方案也没有任何效果。
如果有任何信息或想法，我将非常感激！
endpoint.audDevManager().setNullDev()