我正在使用 JAIN SIP API 的 NIST 实现在 Java 中开发一个 SIP 控制器。
我无法通过 Asterisk 从我的 SIP 控制器拨打软件电话。如果我使用其 IP 地址和端口号直接（而不是通过 Asterisk）呼叫软件电话，一切正常。呼叫建立，软件电话听到我发送的音频（RTP 数据），我可以接收它发送给我的音频。
但是，当我通过 Asterisk 呼叫同一个软件电话时，呼叫建立，并且我开始从软件电话接收 RTP 数据（通过 Asterisk）。现在，我的发送流需要一些时间来设置，但在配置它时，我会从软件电话接收 RTP 数据。问题是，一旦我的发送流被初始化并开始传输 RTP 数据，我就停止从软件电话接收 RTP 数据！结果是通话建立后，我听到软电话半秒或最多一秒，然后什么也听不见。在这个阶段，软电话可以听到我传出的 RTP 数据，但我听不到。
如果我不开始传输任何 RTP 数据，我会继续从软件电话接收 RTP 数据。但是一旦我开始传输，它就停止了！
如果有帮助，这里是建立呼叫的 SIP 对话类型（>> 表示传出消息，<< 表示传入消息）：
>> INVITE sip:301@asterisk SIP/2.0
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4@10.0.85.3
CSeq: 1 INVITE
From: <sip:null>;tag=JqbJKA
To: <sip:301@asterisk>
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bK34d24b3f748ac08a5ca46f500f110d38353436
Max-Forwards: 70
Contact: <sip:10.0.85.3:5060>
Route: <sip:10.0.84.30;lr>
Content-Type: application/sdp
Content-Length: 106
v=0
o=- 3515232260 3515232260 IN IP4 10.0.85.3
s=-
c=IN IP4 10.0.85.3
t=0 0
m=audio 42138 RTP/AVP 0
a=rtpmap:0 PCMU/8000
<< SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bK34d24b3f748ac08a5ca46f500f110d38353436;received=10.0.85.3
From: <sip:null>;tag=JqbJKA
To: <sip:301@asterisk>;tag=as7077f414
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4@10.0.85.3
CSeq: 1 INVITE
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:301@10.0.84.30>
Proxy-Authenticate: Digest realm="asterisk",nonce="4a1cbda4"
Content-Length: 0
>> INVITE sip:301@asterisk SIP/2.0
CSeq: 2 INVITE
From: <sip:303@asterisk>;tag=JqbJKA
To: <sip:301@asterisk>
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKd1870f50e9fbf883b3e64fa3ef75dda9353436
Max-Forwards: 70
Contact: <sip:10.0.85.3:5060>
Route: <sip:10.0.84.30;lr>
Proxy-Authorization: Digest username="303",realm="asterisk",nonce="4a1cbda4",response="249b2b7d7c0e7b54499c632ba410365c",algorithm=MD5,uri="sip:301@asterisk",nc=00000001
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4@10.0.85.3
Content-Type: application/sdp
Content-Length: 106
v=0
o=- 3515232260 3515232260 IN IP4 10.0.85.3
s=-
c=IN IP4 10.0.85.3
t=0 0
m=audio 42138 RTP/AVP 0
a=rtpmap:0 PCMU/8000`
`<< SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKd1870f50e9fbf883b3e64fa3ef75dda9353436;received=10.0.85.3
From: <sip:303@asterisk>;tag=JqbJKA
To: <sip:301@asterisk>
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4@10.0.85.3
CSeq: 2 INVITE
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,R EFER,SUBSCRIBE,NOTIFY
Contact: <sip:301@10.0.84.30>
Content-Length: 0
`<< SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKd1870f50e9fbf883b3e64fa3ef75dda9353436;received=10.0.85.3
From: <sip:303@asterisk>;tag=JqbJKA
To: <sip:301@asterisk>;tag=as00faa25e
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4@10.0.85.3
CSeq: 2 INVITE
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:301@10.0.84.30>
Content-Length: 0`
<< SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKd1870f50e9fbf883b3e64fa3ef75dda9353436;received=10.0.85.3
From: <sip:303@asterisk>;tag=JqbJKA
To: <sip:301@asterisk>;tag=as00faa25e
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4@10.0.85.3
CSeq: 2 INVITE
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:301@10.0.84.30>
Content-Type: application/sdp
Content-Length: 154
v=0
o=root 2593 2593 IN IP4 10.0.84.30
s=session
c=IN IP4 10.0.84.30
t=0 0
m=audio 10294 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
>> ACK sip:301@10.0.84.30 SIP/2.0
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4@10.0.85.3
CSeq: 2 ACK
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bK7e16ebc0de9c6eaf901db0e2e58f495f353436
From: <sip:303@asterisk>;tag=JqbJKA
To: <sip:301@asterisk>;tag=as00faa25e
Max-Forwards: 70
Contact: <sip:10.0.85.3:5060>
Content-Length: 0
这是设置 RTP 会话的代码。首先是一些声明：
private RTPManager sessionManager = null;
private Processor processor = null;
private SendStream sendStream;`
首先调用以下方法：
public void startMedia(String peerIp,int peerPort,int receivePort,String format) throws IOException,MediaException,InvalidSessionAddressException
{
stopMedia();
this.format = format;
RTPSessionMgr rtpSessionMgr = new RTPSessionMgr();
rtpSessionMgr.initSession(new SessionAddress(),null,0.05,0.25);
InetAddress localhost = InetAddress.getLocalHost();
SessionAddress localAddr = new SessionAddress(localhost,receivePort,localhost,receivePort + 1);
InetAddress destAddr = InetAddress.getByName(peerIp);
rtpSessionMgr.startSession(localAddr,localAddr,new SessionAddress(destAddr,peerPort,destAddr,peerPort + 1),null);
sessionManager = rtpSessionMgr;
for (ReceiveStreamListener nextListener : receiveStreamListeners)
sessionManager.addReceiveStreamListener(nextListener);
}
然后，要开始通过 RTP 播放声音，调用此方法：
public void transmitSound(DataSource ds) throws NoProcessorException,IOException,UnsupportedFormatException,NotRealizedError
{
stopTransmittingSound();
processor = Manager.createProcessor(ds);
for (ControllerListener nextListener : controllerListeners)
processor.addControllerListener(nextListener);
processor.addControllerListener(myControllerListener);
processor.configure();
}
这是控制器侦听器的 controllerUpdate() 方法：
public void controllerUpdate(ControllerEvent event)
{
if (processor.getState()==Processor.Configured)
{
processor.setContentDescriptor(new ContentDescriptor(ContentDescriptor.RAW_RTP));
processor.getTrackControls()[0].setFormat(new AudioFormat(format,8000,8,1));
processor.realize();
}
else if (processor.getState()==Processor.Realized)
{
try
{
sendStream = sessionManager.createSendStream(processor.getDataOutput(),0);
sendStream.start();
processor.start();
}
catch (IOException e)
{
e.printStackTrace();
}
catch (UnsupportedFormatException e)
{
e.printStackTrace();
}
catch (NotRealizedError e)
{
e.printStackTrace();
}
}
}
这基本上是发送 ACK 后发生的情况：
- 我创建了一个用于传输和收听的 RTP 会话。
- 我开始初始化一个用于传输 RTP 的处理器。
- 与此同时，我收到了大量的 RTP 数据。
- 处理器完成初始化，我开始发送 RTP 数据。
- 在这个阶段，如果通过 Asterisk，我将停止接收 RTP 数据。如果直接拨打软电话，一切正常。
有任何想法吗？