为什么我的 asterisk 服务器没有收到带有以下扭曲 SIP python 代码的响应
Why am I not getting a response from my asterisk server with the following twisted SIP python code
使用此处找到的代码 https://github.com/twisted/twisted/blob/trunk/src/twisted/test/test_sip.py
我已经尝试向我的本地 (127.0.0.1) 星号服务器发送简单的 SIP 选项检查。
From twisted.protocols import sip
From twisted.internet import defer, reactor
class Client(sip.Base):
def __init__(self):
sip.Base.__init__(self)
self.received = []
self.deferred = defer.Deferred()
def handle_response(self, response, addr):
self.received.append(response)
self.deferred.callback(self.received)
class OptionsC():
def setup(self):
self.client = Client()
self.clientPort = reactor.listenUDP(
5062, self.client, interface="127.0.1.1")
def testRegisterOPT(self):
p = self.clientPort.getHost().port
r = sip.Request("OPTIONS", "sip:127.0.0.1")
r.addHeader("via", sip.Via("127.0.1.1", port=5062, rport=5062, branch="test123").toString())
r.addHeader("to", "<sip:joe@127.0.1.1>")
r.addHeader("From", "<sip:joe@127.0.1.1>")
r.addHeader("Call-ID", "<1opt@127.0.1.1>")
r.addHeader("CSeq", "1 OPTIONS")
r.addHeader("contact", "<sip:joe@127.0.1.1:5062;transport=UDP>")
r.addHeader("Accept", "application/sdp")
r.addHeader("Content-Length", "0")
print(r.toString())
self.client.sendMessage(
sip.URL(host="127.0.0.1", port=5060), r)
d = self.client.deferred
def check(received):
self.assertEqual(len(received), 1)
r = received[0]
print(r)
print(r.code)
print(dir(r))
self.assertEqual(r.code, 200)
d.addCallback(check)
return d
opc = OptionsC()
opc.setup()
res = opc.testRegisterOPT()
print("test")
这就是我的环境
$ pip freeze
constantly==15.1.0
incremental==16.10.1
Twisted==16.6.0
zope.interface==4.3.3
$ python -V
Python 2.7.5
在星号上，我可以看到消息到达。
但是星号从不发送响应 200 ok，或错误或任何其他对我理解我的代码有什么问题有用的东西。
*CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:127.0.1.1:5062 --->
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5062;branch=test123;rport=5062
To: <sip:joe@127.0.1.1>
From: <sip:joe@127.0.1.1>
Call-ID: <1opt@127.0.1.1>
CSeq: 1 OPTIONS
Contact: <sip:joe@127.0.1.1:5062;transport=UDP>
Accept: application/sdp
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
将此与下面的 sipp 命令进行比较，后者的功能非常相似。
$ sipp -sf ./options.xml -m 1 -max_non_invite_retrans 1 127.0.0.1:5060
Resolving remote host '127.0.0.1'... Done.
$ cat ./options.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Options">
<send retrans="500" start_rtd="opt-timer">
<![CDATA[
OPTIONS sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
To: <sip:[service]@[remote_ip]>
From: <sip:[service]@[remote_ip]>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] OPTIONS
Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
Accept: application/sdp
Content-Length: 0
]]>
</send>
<recv response="200" rrs="true" rtd="opt-timer"></recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
哪个星号正确地产生了对
的响应
<--- SIP read from UDP:127.0.0.1:5061 --->
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5061;branch=z9hG4bK-19903-1-0
Max-Forwards: 70
To: <sip:service@127.0.0.1>
From: <sip:service@127.0.0.1>;tag=1
Call-ID: 1-19903@127.0.1.1
CSeq: 1 OPTIONS
Contact: <sip:service@127.0.1.1:5061;transport=UDP>
Accept: application/sdp
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 127.0.0.1:5061 (NAT)
Looking for s in public (domain 127.0.0.1)
<--- Transmitting (NAT) to 127.0.0.1:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.1.1:5061;branch=z9hG4bK-19903-1-0;received=127.0.0.1;rport=5061
From: <sip:service@127.0.0.1>;tag=1
To: <sip:service@127.0.0.1>;tag=as6e9328e8
Call-ID: 1-19903@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 13.10.0~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:127.0.0.1:5060>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1-19903@127.0.1.1' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '1-19903@127.0.1.1' Method: OPTIONS
我看到的是 SIP 正在后台做一些其他工作来初始化 SIP 对话。
我认为这是上面的示例代码中缺少的。
当 asterisk 正确启动传输到客户端 sipp 正在创建的响应时。
删除 r.addHeader("CSeq", "1 OPTIONS").
我现在正在让我的本地星号响应
def testRegisterOPT(self):
p = self.clientPort.getHost().port
r = sip.Request("OPTIONS", "sip:127.0.0.1")
r.addHeader("via", sip.Via("127.0.1.1", port=5062, rport=5062, branch="test123").toString())
r.addHeader("to", "<sip:joe@127.0.1.1>")
r.addHeader("From", "<sip:joe@127.0.1.1>")
r.addHeader("Call-ID", "<1opt@127.0.1.1>")
# r.addHeader("CSeq", "1 OPTIONS")
r.addHeader("contact", "<sip:joe@127.0.1.1:5062;transport=UDP>")
r.addHeader("Accept", "application/sdp")
r.addHeader("Content-Length", "0")
print(r.toString())
self.client.sendMessage(
sip.URL(host="127.0.0.1", port=5060), r)
d = self.client.deferred
所以我想我需要建立一种方法来正确处理调用序列或忽略它。
有星号张贴的警告。
chan_sip.c:11681 copy_header: No field 'CSeq' present to copy
但至少我的测试现在有效
<--- SIP read from UDP:127.0.0.1:46947 --->
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062
Max-Forward: 3
To: <sip:OptionsCheckMonitor@127.0.0.1>
From: <sip:OptionsCheckMonitor@127.0.0.1>
Call-ID: <1optioncheck@127.0.0.1>
Contact: <sip:OptionsCheckMonitor@127.0.0.1;transport=UDP>
Accept: application/sdp
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 127.0.0.1:5062 (no NAT)
[Jan 24 12:52:54] NOTICE[12752]: chan_sip.c:11681 copy_header: No field 'CSeq' present to copy
<--- Transmitting (no NAT) to 127.0.0.1:5062 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;received=127.0.0.1
From: <sip:OptionsCheckMonitor@127.0.0.1>
To: <sip:OptionsCheckMonitor@127.0.0.1>;tag=as7d34b365
Call-ID: <1optioncheck@127.0.0.1>
Server: Asterisk PBX 13.10.0~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
使用此处找到的代码 https://github.com/twisted/twisted/blob/trunk/src/twisted/test/test_sip.py
我已经尝试向我的本地 (127.0.0.1) 星号服务器发送简单的 SIP 选项检查。
From twisted.protocols import sip
From twisted.internet import defer, reactor
class Client(sip.Base):
def __init__(self):
sip.Base.__init__(self)
self.received = []
self.deferred = defer.Deferred()
def handle_response(self, response, addr):
self.received.append(response)
self.deferred.callback(self.received)
class OptionsC():
def setup(self):
self.client = Client()
self.clientPort = reactor.listenUDP(
5062, self.client, interface="127.0.1.1")
def testRegisterOPT(self):
p = self.clientPort.getHost().port
r = sip.Request("OPTIONS", "sip:127.0.0.1")
r.addHeader("via", sip.Via("127.0.1.1", port=5062, rport=5062, branch="test123").toString())
r.addHeader("to", "<sip:joe@127.0.1.1>")
r.addHeader("From", "<sip:joe@127.0.1.1>")
r.addHeader("Call-ID", "<1opt@127.0.1.1>")
r.addHeader("CSeq", "1 OPTIONS")
r.addHeader("contact", "<sip:joe@127.0.1.1:5062;transport=UDP>")
r.addHeader("Accept", "application/sdp")
r.addHeader("Content-Length", "0")
print(r.toString())
self.client.sendMessage(
sip.URL(host="127.0.0.1", port=5060), r)
d = self.client.deferred
def check(received):
self.assertEqual(len(received), 1)
r = received[0]
print(r)
print(r.code)
print(dir(r))
self.assertEqual(r.code, 200)
d.addCallback(check)
return d
opc = OptionsC()
opc.setup()
res = opc.testRegisterOPT()
print("test")
这就是我的环境
$ pip freeze
constantly==15.1.0
incremental==16.10.1
Twisted==16.6.0
zope.interface==4.3.3
$ python -V
Python 2.7.5
在星号上，我可以看到消息到达。
但是星号从不发送响应 200 ok，或错误或任何其他对我理解我的代码有什么问题有用的东西。
*CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:127.0.1.1:5062 --->
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5062;branch=test123;rport=5062
To: <sip:joe@127.0.1.1>
From: <sip:joe@127.0.1.1>
Call-ID: <1opt@127.0.1.1>
CSeq: 1 OPTIONS
Contact: <sip:joe@127.0.1.1:5062;transport=UDP>
Accept: application/sdp
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
将此与下面的 sipp 命令进行比较，后者的功能非常相似。
$ sipp -sf ./options.xml -m 1 -max_non_invite_retrans 1 127.0.0.1:5060
Resolving remote host '127.0.0.1'... Done.
$ cat ./options.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Options">
<send retrans="500" start_rtd="opt-timer">
<![CDATA[
OPTIONS sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
To: <sip:[service]@[remote_ip]>
From: <sip:[service]@[remote_ip]>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] OPTIONS
Contact: <sip:[service]@[local_ip]:[local_port];transport=[transport]>
Accept: application/sdp
Content-Length: 0
]]>
</send>
<recv response="200" rrs="true" rtd="opt-timer"></recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
哪个星号正确地产生了对
的响应<--- SIP read from UDP:127.0.0.1:5061 --->
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5061;branch=z9hG4bK-19903-1-0
Max-Forwards: 70
To: <sip:service@127.0.0.1>
From: <sip:service@127.0.0.1>;tag=1
Call-ID: 1-19903@127.0.1.1
CSeq: 1 OPTIONS
Contact: <sip:service@127.0.1.1:5061;transport=UDP>
Accept: application/sdp
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 127.0.0.1:5061 (NAT)
Looking for s in public (domain 127.0.0.1)
<--- Transmitting (NAT) to 127.0.0.1:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.1.1:5061;branch=z9hG4bK-19903-1-0;received=127.0.0.1;rport=5061
From: <sip:service@127.0.0.1>;tag=1
To: <sip:service@127.0.0.1>;tag=as6e9328e8
Call-ID: 1-19903@127.0.1.1
CSeq: 1 OPTIONS
Server: Asterisk PBX 13.10.0~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:127.0.0.1:5060>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1-19903@127.0.1.1' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '1-19903@127.0.1.1' Method: OPTIONS
我看到的是 SIP 正在后台做一些其他工作来初始化 SIP 对话。 我认为这是上面的示例代码中缺少的。 当 asterisk 正确启动传输到客户端 sipp 正在创建的响应时。
删除 r.addHeader("CSeq", "1 OPTIONS").
我现在正在让我的本地星号响应
def testRegisterOPT(self):
p = self.clientPort.getHost().port
r = sip.Request("OPTIONS", "sip:127.0.0.1")
r.addHeader("via", sip.Via("127.0.1.1", port=5062, rport=5062, branch="test123").toString())
r.addHeader("to", "<sip:joe@127.0.1.1>")
r.addHeader("From", "<sip:joe@127.0.1.1>")
r.addHeader("Call-ID", "<1opt@127.0.1.1>")
# r.addHeader("CSeq", "1 OPTIONS")
r.addHeader("contact", "<sip:joe@127.0.1.1:5062;transport=UDP>")
r.addHeader("Accept", "application/sdp")
r.addHeader("Content-Length", "0")
print(r.toString())
self.client.sendMessage(
sip.URL(host="127.0.0.1", port=5060), r)
d = self.client.deferred
所以我想我需要建立一种方法来正确处理调用序列或忽略它。
有星号张贴的警告。
chan_sip.c:11681 copy_header: No field 'CSeq' present to copy
但至少我的测试现在有效
<--- SIP read from UDP:127.0.0.1:46947 --->
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062
Max-Forward: 3
To: <sip:OptionsCheckMonitor@127.0.0.1>
From: <sip:OptionsCheckMonitor@127.0.0.1>
Call-ID: <1optioncheck@127.0.0.1>
Contact: <sip:OptionsCheckMonitor@127.0.0.1;transport=UDP>
Accept: application/sdp
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 127.0.0.1:5062 (no NAT)
[Jan 24 12:52:54] NOTICE[12752]: chan_sip.c:11681 copy_header: No field 'CSeq' present to copy
<--- Transmitting (no NAT) to 127.0.0.1:5062 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;received=127.0.0.1
From: <sip:OptionsCheckMonitor@127.0.0.1>
To: <sip:OptionsCheckMonitor@127.0.0.1>;tag=as7d34b365
Call-ID: <1optioncheck@127.0.0.1>
Server: Asterisk PBX 13.10.0~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0