"無會話，非5G”鏈接如下：https://mp.weixin.qq.com/s/AX3MCn1woaSRrDVd2nkrtA
在VOLTE學(xué)習(xí)中你可能曾聽說過“鬼振鈴"(ghost rings）,其具體在RFC 3312中有專門的解釋和說明；”鬼振鈴”在VOLTE通話具體表現(xiàn)就是被叫的電話接通，但是聽不到主叫的聲音！
1. 鬼振鈴(ghost rings）產(chǎn)生原因
出現(xiàn)“鬼振鈴”這種情況的原因是緣于通信網(wǎng)絡(luò)無論是數(shù)據(jù)或話音都是遵循“先進(jìn)行控制面（信令）鏈路的搭建，再進(jìn)行用戶面（數(shù)據(jù)或語音鏈路）分配”原則；因此，在控制面建立成功后就需要等待用戶面承載（或鏈路）的分配，期間（電話）振鈴后資源未分配到位，被叫聽不到對方通話的情況；下圖所示：
09:00 用戶A約用戶B討論問題，用戶B答應(yīng)30分鐘后在301會議室碰頭；
09:25 用戶B告訴用戶A，他有其他約會，301會議室有人使用！
問題原因：用戶B未安排好約會與會議時間，之前未確認(rèn)301會議室09:30分后是否有人使用或預(yù)定；
2.提前預(yù)留（precondition）
為了避免出現(xiàn)“鬼振鈴"(ghost rings）影響用戶感知，在 RFC 3312 協(xié)議中設(shè)計了Precodition（提前預(yù)留），也就是呼叫發(fā)出后用戶面進(jìn)行承載分配預(yù)留（RESERVATION)；VOLTE通話中的Precodition（提前預(yù)留）過程見下圖：
3.precondition指示
用戶面承載預(yù)留（RESERVATION)分別由主、被與CSCF進(jìn)行交互，通過INVITE等消息指示，具體見以下消息中紅色字體部分：
3.1 INVITE
INVITE sip:user1@192.168.1.11:37607 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:64700;branch=z9hG4bK635101086023655710;rport;transport=UDP
Max-Forwards: 69
Call-ID: [email]635101086023565705@192.168.1.2[/email]
CSeq: 6 INVITE
Via: SIP/2.0/UDP 192.168.1.2:64627;branch=z9hG4bK635101086023565705;transport=UDP
To: <sip:user1@anritsu-cscf.com>
From: <sip:0123456789@anritsu-cscf.com>;tag=1111111111
Contact: <sip:0123456789@anritsu-cscf.com:64627;transport=udp>
Allow: INVITE, UPDATE, ACK, CANCEL, BYE, PRACK, MESSAGE
Supported: 100rel // Indicate "PRACK" is supported
Supported: precondition // Indicate "Precondition" is supported
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-application.ims.iari.gsma-vs";require;explicit
Privacy: id
Content-Type: application/sdp
Content-Length: 565
Record-Route: <sip:192.168.1.2;lr>
3.2 183 Session Progress
Max-Forwards: 70
Via: SIP/2.0/TCP [2001:0:0:2::1]:5060;branch=z9hG4bK932432170smg;transport=TCP
From: <sip:310410123456789@test.net>;tag=2763466811
To: <sip:0123456789;phone-context=test.net@test.net;user=phone>;tag=1111111111
Call-ID: 2270680280
CSeq: 1 INVITE
Contact: <sip:0123456789@[2001:0:0:2::2]:65094;transport=tcp>
Record-Route: <sip:[2001:0:0:2::2];lr>
Content-Type: application/sdp
Require: precondition
Require: 100rel
RSeq: 1
Content-Length: 763
Privacy: none
P-Asserted-Identity: <sip:0123456789>;phone-context=test.net@test.net;user=phone
v=0
o=anritsu 1 1 IN IP6 2001:0:0:2::2
s=SS VOIP
c=IN IP6 2001:0:0:2::2
t=0 0
m=audio 60000 RTP/AVP 116 107 97 115 111 110
b=AS:49
a=rtpmap:116 AMR-WB/16000
a=fmtp:116 mode-change-capability=2; max-red=0
a=rtpmap:107 AMR-WB/16000
a=fmtp:107 octet-align=1;mode-change-capability=2; max-red=0
a=rtpmap:97 AMR/8000
a=fmtp:97 mode-change-capability=2; max-red=0
a=rtpmap:115 AMR/8000
a=fmtp:115 octet-align=1;mode-change-capability=2; max-red=0
a=rtpmap:111 telephone-event/16000
a=fmtp:111 0-15
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
a=curr:qos local none // Indicate 'local' is not ready for communication since it does not meet 'des' condition
a=curr:qos remote none // Indicate 'remote' is not ready for communication since it does not meet 'des' condition
a=des:qos mandatory local sendrecv
a=des:qos mandatory remote sendrecv
a=sendrecv
a=ptime:20
a=maxptime:240
a=conf:qos remote sendrecv
a=rtcp:60001
3.3 200 OK
Max-Forwards: 70
Via: SIP/2.0/TCP [2001:0:0:2::1]:5060;branch=z9hG4bK1835967788smg;transport=TCP
From: <sip:310410123456789@test.net>;tag=2763466811
To: <sip:0123456789;phone-context=test.net@test.net;user=phone>;tag=1111111111
Call-ID: 2270680280
CSeq: 3 UPDATE
Allow: INVITE, ACK, CANCEL, BYE, PRACK, MESSAGE
Contact: <sip:0123456789@[2001:0:0:2::2]:65094;transport=tcp>
Record-Route: <sip:[2001:0:0:2::2];lr>
Content-Type: application/sdp
Content-Length: 743
Privacy: none
v=0
o=anritsu 1 2 IN IP6 2001:0:0:2::2
s=SS VOIP
c=IN IP6 2001:0:0:2::2
t=0 0
m=audio 60000 RTP/AVP 116 107 97 115 111 110
b=AS:49
a=rtpmap:116 AMR-WB/16000
a=fmtp:116 mode-change-capability=2; max-red=0
a=rtpmap:107 AMR-WB/16000
a=fmtp:107 octet-align=1;mode-change-capability=2; max-red=0
a=rtpmap:97 AMR/8000
a=fmtp:97 mode-change-capability=2; max-red=0
a=rtpmap:115 AMR/8000
a=fmtp:115 octet-align=1;mode-change-capability=2; max-red=0
a=rtpmap:111 telephone-event/16000
a=fmtp:111 0-15
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
a=curr:qos local sendrecv // Indicate 'local' is ready for communication since it meets 'des' condition
a=curr:qos remote sendrecv // Indicate 'remote' is ready for communication since it meets 'des' condition
a=des:qos mandatory local sendrecv
a=des:qos mandatory remote sendrecv
a=sendrecv
a=ptime:20
a=maxptime:240
a=rtcp:60001
原文來自：http://www.sharetechnote.com/
整理：kangguoying20200929
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