我希望這將是我對這個SIP主題的最後一個問題,我設法克服了最後一個issue我通過讓朋友幫助我從遠程計算機,我可以在計算機之間進行連接,但根據我看到的所有examples,被調用者都應該調用Ringing響應,但在我的應用程序中,我還沒有實現它,但我仍然收到在主叫方UAC一個振鈴響應,這是在呼叫者端的SIP消息:在SIP UAC上獲得RINGING響應,而不從其他UAC發送它
呼出請求5:
INVITE sip:[email protected] SIP/2.0
Contact: "Client 310" <sip:[email protected]>
From: "Client 310" <sip:[email protected]>
Max-Forwards: 32
CSeq: 2 INVITE
Call-ID: [email protected]
Allow: INVITE,CANCEL,ACK,BYE,OPTIONS
Content-Type: application/sdp
Proxy-Authorization: Digest username="310",nonce="012afffb",realm="asterisk",uri="sip:[email protected]",algorithm=MD5,response="d19ca5b98450b4be7bd4045edb8a3a2f"
Via: SIP/2.0/UDP hostName.hn:5060
To: "Client 320" <sip:[email protected]>;tag=as5a8fa200
Content-Length: 257
v=0
o=310 7108915969559970847 7108915969559970847 IN IP4 xxx.xxx.x.xxx
s=-
i=Nu-Art Software - TacB0sS VoIP information
c=IN IP4 xxx.xxx.x.xxx
m=audio 3312 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
傳入響應6:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233
From: "Client 310" <sip:[email protected]>
To: "Client 320" <sip:[email protected]>;tag=as5a8fa200
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0
傳入響應7:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233
From: "Client 310" <sip:[email protected]>
To: "Client 320" <sip:[email protected]>;tag=as5a8fa200
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0
電話:[email protected]正在振鈴
傳入響應8 :
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233
From: "Client 310" <sip:[email protected]>
To: "Client 320" <sip:[email protected]>;tag=as5a8fa200
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 264
v=0
o=root 27669 27669 IN IP4 yy.yy.yy.yy
s=session
c=IN IP4 yy.yy.yy.yy
t=0 0
m=audio 10914 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
傳入響應9:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233
From: "Client 310" <sip:[email protected]>
To: "Client 320" <sip:[email protected]>;tag=as5a8fa200
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Freeswitch 1.2.3
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces
Content-Length: 0
我不迴應的邀請,這就是爲什麼這一切正在發生的事情,但爲什麼我得到一個響,如果我不是一個發送它。
謝謝,
亞當。
更新:
如果你會發現,我得到的響應時間:
傳入響應7: 振鈴
傳入響應8: 會話進度
傳入響應9: 服務不可用
我不明白這裏的邏輯,我從第一振鈴的會話進度8秒,但是從會話進程到服務不可用我有47ms?
這有什麼意義? 50毫秒做什麼?時間來分析響應+打開RTP會話的時間+構建響應的時間+構建SDP +的時間服務器接收消息所需的時間 - 503消息到達我的UAC所需的時間不是這個切割點很近?在這一點上,我想回應服務器?再次
感謝您的所有幫助奇才。
生成的音?渲染? 我並不熟悉這些術語...... – TacB0sS 2010-06-07 01:26:48
您的意思是開始發送RTP數據包? – TacB0sS 2010-06-07 01:35:49
生成是指當SIP設備收到沒有RTP的特定響應(如180振鈴(180振鈴也可以有RTP,但它是可選的))時將播放的音頻。渲染是指當SIP設備具有可用RTP流時將播放的音頻。生成和渲染不是正式的術語,而是我用來解釋發生了什麼的描述性術語。 – sipwiz 2010-06-07 01:35:55