Question

我正在写一个SIP服务器，我接听电话，然后把它们连接到一部voip电话，问题是当你挂掉voip电话时，转发BYE信息的地方有问题我的手机没有结束通话。

这是SIP消息日志（我用1234替换了服务器的电话号码，用5678替换了我的手机电话号码，我的服务器的IP已被x替换，我的voip电话的IP已被y替换） -

Incoming from 174.37.45.134:5060 - 
INVITE sip:1234@174.37.45.134:5060;rinstance=f10c56ae7fb62958 SIP/2.0
Record-Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>
Record-Route: <sip:67.228.177.9;lr=on;ftag=VPSF506071629460>
Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>
Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>
Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bK9767.ad406992.0
Via: SIP/2.0/UDP 67.228.177.9;rport=5060;branch=z9hG4bK9767.760c9624.0
Via: SIP/2.0/UDP 216.82.224.202;rport=5060;received=216.82.224.202;branch=z9hG4bK9767.823f8e12.0
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK9767.723f8e12.0
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK9767.e30c5303.0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1256581032616
f: "Carro Ramon"  <sip:5678@4.68.250.148>;tag=VPSF506071629460
t: <sip:+11234@4.79.212.229:5060>
i: MIAMGC0120091027172219041244@209.244.63.32
CSeq: 1 INVITE
m: <sip:+15678@4.68.250.148:5060;transport=udp;nat=yes>
Max-Forwards: 64
c: application/sdp
Content-Length: 192

v=0
o=- 1256664139 1256664140 IN IP4 209.247.22.135
s=-
c=IN IP4 174.37.45.134
t=0 0
m=audio 55540 RTP/AVP 0 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes

Outgoing to 174.37.45.134:5060 - 
SIP/2.0 180 Ringing
CSeq: 1 INVITE
Call-ID: MIAMGC0120091027172219041244@209.244.63.32
Contact: <sip:+15678@4.68.250.148:5060;transport=udp;nat=yes>
From: "Carro Ramon"  <sip:5678@4.68.250.148>;tag=VPSF506071629460
Max-Forwards: 70
Record-Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>, <sip:67.228.177.9;lr=on;ftag=VPSF506071629460>, <sip:216.82.224.202;lr;ftag=VPSF506071629460>, <sip:4.79.212.229;lr;ftag=VPSF506071629460>
To: <sip:+11234@4.79.212.229:5060>;tag=dAmXcBGL
Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bK9767.ad406992.0, SIP/2.0/UDP 67.228.177.9;rport=5060;branch=z9hG4bK9767.760c9624.0, SIP/2.0/UDP 216.82.224.202;rport=5060;received=216.82.224.202;branch=z9hG4bK9767.823f8e12.0, SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK9767.723f8e12.0, SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK9767.e30c5303.0, SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1256581032616
Content-Length: 0

Outgoing to 174.37.45.134:5060 - 
SIP/2.0 200 OK
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
CSeq: 1 INVITE
Call-ID: MIAMGC0120091027172219041244@209.244.63.32
Contact: <sip:+15678@4.68.250.148:5060;transport=udp;nat=yes>
Content-Type: application/sdp
From: "Carro Ramon"  <sip:5678@4.68.250.148>;tag=VPSF506071629460
Max-Forwards: 70
Record-Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>, <sip:67.228.177.9;lr=on;ftag=VPSF506071629460>, <sip:216.82.224.202;lr;ftag=VPSF506071629460>, <sip:4.79.212.229;lr;ftag=VPSF506071629460>
To: <sip:+11234@4.79.212.229:5060>;tag=BYFeP7T1
Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bK9767.ad406992.0, SIP/2.0/UDP 67.228.177.9;rport=5060;branch=z9hG4bK9767.760c9624.0, SIP/2.0/UDP 216.82.224.202;rport=5060;received=216.82.224.202;branch=z9hG4bK9767.823f8e12.0, SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK9767.723f8e12.0, SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK9767.e30c5303.0, SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1256581032616
Content-Length: 206

v=0
o=Zoiper_user 0 0 IN IP4 xx.xx.xxx.xx
s=Zoiper_session
c=IN IP4 xx.xx.xxx.xx
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Incoming from 174.37.45.134:5060 - 
ACK sip:+15678@xx.xx.xxx.xx:5060;transport=udp SIP/2.0
Record-Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>
Record-Route: <sip:67.228.177.9;lr=on;ftag=VPSF506071629460>
Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>
Via: SIP/2.0/UDP 174.37.45.134;branch=z9hG4bK9767.ad406992.2
Via: SIP/2.0/UDP 67.228.177.9;rport=5060;branch=z9hG4bK9767.760c9624.2
Via: SIP/2.0/UDP 216.82.224.202;rport=5060;received=216.82.224.202;branch=z9hG4bK9767.723f8e12.2
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK9767.e30c5303.2
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1256581032653
From: "CARRO RAMON    "  <sip:+15678@4.68.250.148;isup-oli=0>;tag=VPSF506071629460
To: <sip:+11234@4.79.212.229:5060>;tag=BYFeP7T1
Call-ID: MIAMGC0120091027172219041244@209.244.63.32
CSeq: 1 ACK
Contact: <sip:4.68.250.148:5060;transport=udp>
Max-Forwards: 66
Content-Length: 0

Outgoing to yyy.yyy.yy.yyy:1024 - 
INVITE sip:3998@192.168.1.121 SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
CSeq: 1 INVITE
Call-ID: AW6zfKQ8RWl71MipIe4X1WWKfw7xGGR9@chat.seohosting.com
Contact: <sip:5678@xx.xx.xxx.xx>;transport=UDP
Content-Type: application/sdp
From: "(null)" <sip:5678@xx.xx.xxx.xx>;transport=UDP;tag=7b2add35
Max-Forwards: 70
To: <sip:3998@xx.xx.xxx.xx>
User-Agent: Zoiper rev.4186
Via: SIP/2.0/UDP xx.xx.xxx.xx:5060
Content-Length: 206

v=0
o=Zoiper_user 0 0 IN IP4 xx.xx.xxx.xx
s=Zoiper_session
c=IN IP4 xx.xx.xxx.xx
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Incoming from yyy.yyy.yy.yyy:1024 - 
SIP/2.0 100 Trying
To: <sip:3998@xx.xx.xxx.xx>
From: "(null)" <sip:5678@xx.xx.xxx.xx>;transport=UDP;tag=7b2add35
Call-ID: AW6zfKQ8RWl71MipIe4X1WWKfw7xGGR9@chat.seohosting.com
CSeq: 1 INVITE
Via: SIP/2.0/UDP xx.xx.xxx.xx:5060
Server: Linksys/SPA941-5.1.8
Content-Length: 0


Incoming from yyy.yyy.yy.yyy:1024 - 
SIP/2.0 180 Ringing
To: <sip:3998@xx.xx.xxx.xx>;tag=53cca4372c533924i0
From: "(null)" <sip:5678@xx.xx.xxx.xx>;transport=UDP;tag=7b2add35
Call-ID: AW6zfKQ8RWl71MipIe4X1WWKfw7xGGR9@chat.seohosting.com
CSeq: 1 INVITE
Via: SIP/2.0/UDP xx.xx.xxx.xx:5060
Server: Linksys/SPA941-5.1.8
Content-Length: 0

Incoming from yyy.yyy.yy.yyy:1024 - 
SIP/2.0 200 OK
To: <sip:3998@xx.xx.xxx.xx>;tag=53cca4372c533924i0
From: "(null)" <sip:5678@xx.xx.xxx.xx>;transport=UDP;tag=7b2add35
Call-ID: AW6zfKQ8RWl71MipIe4X1WWKfw7xGGR9@chat.seohosting.com
CSeq: 1 INVITE
Via: SIP/2.0/UDP xx.xx.xxx.xx:5060
Contact: "3998" <sip:3998@192.168.1.121:5060>
Server: Linksys/SPA941-5.1.8
Content-Length: 212
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 49591664 49591664 IN IP4 192.168.1.121
s=-
c=IN IP4 192.168.1.121
t=0 0
m=audio 16432 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

Outgoing to yyy.yyy.yy.yyy:1024 - 
ACK sip:3998@192.168.1.121 SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
CSeq: 1 ACK
Call-ID: AW6zfKQ8RWl71MipIe4X1WWKfw7xGGR9@chat.seohosting.com
Contact: <sip:5678@xx.xx.xxx.xx>;transport=UDP
From: "(null)" <sip:5678@xx.xx.xxx.xx>;transport=UDP;tag=7b2add35
Max-Forwards: 70
To: <sip:3998@xx.xx.xxx.xx>;tag=53cca4372c533924i0
User-Agent: Zoiper rev.4186
Via: SIP/2.0/UDP xx.xx.xxx.xx:5060
Content-Length: 0

Incoming from yyy.yyy.yy.yyy:1024 - 
BYE sip:5678@xx.xx.xxx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-598f1319
From: <sip:3998@xx.xx.xxx.xx>;tag=53cca4372c533924i0
To: "(null)" <sip:5678@xx.xx.xxx.xx>;tag=7b2add35
Call-ID: AW6zfKQ8RWl71MipIe4X1WWKfw7xGGR9@chat.seohosting.com
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 0

Outgoing to 174.37.45.134:5060 - 
BYE sip:5678@4.68.250.148 SIP/2.0
CSeq: 2 BYE
Call-ID: MIAMGC0120091027172219041244@209.244.63.32
Contact: <sip:1234@xx.xx.xxx.xx>
From: <sip:+11234@4.79.212.229:5060>;tag=BYFeP7T1
Max-Forwards: 70
Route: <sip:174.37.45.134;lr=on;ftag=VPSF506071629460>, <sip:67.228.177.9;lr=on;ftag=VPSF506071629460>, <sip:216.82.224.202;lr;ftag=VPSF506071629460>
To: "CARRO RAMON    "  <sip:+15678@4.68.250.148;isup-oli=0>;tag=VPSF506071629460
Via: SIP/2.0/UDP 174.37.45.134:5060
Content-Length: 0

Outgoing to yyy.yyy.yy.yyy:1024 - 
SIP/2.0 200 OK
CSeq: 101 BYE
Call-ID: AW6zfKQ8RWl71MipIe4X1WWKfw7xGGR9@chat.seohosting.com
From: <sip:3998@xx.xx.xxx.xx>;tag=53cca4372c533924i0;tag=D1EASwOG
Max-Forwards: 70
To: "(null)" <sip:5678@xx.xx.xxx.xx>;tag=7b2add35
Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-598f1319

Incoming from 174.37.45.134:5060 - 
SIP/2.0 408 Request Timeout
CSeq: 2 BYE
Call-ID: MIAMGC0120091027172219041244@209.244.63.32
From: <sip:+11234@4.79.212.229:5060>;tag=BYFeP7T1
To: "CARRO RAMON    "  <sip:+15678@4.68.250.148;isup-oli=0>;tag=VPSF506071629460
Via: SIP/2.0/UDP 174.37.45.134:5060;rport=5060;received=xx.xx.xxx.xx
Server: Kamailio (1.5.2-notls (x86_64/linux))
Content-Length: 0
Warning: 392 67.228.177.9:5060 "Noisy feedback tells:  pid=15004 req_src_ip=174.37.45.134 req_src_port=5060 in_uri=sip:5678@4.68.250.148 out_uri=sip:5678@4.68.250.148 via_cnt==1092"

Answer 1

您可能想要检查警告标题的值是什么。有一些自定义消息“噪声反馈告诉”...这是非常特定于应用程序。当事务超时到期时，请求超时消息通常由堆栈模拟。这可能意味着您对174.37.45.134:5060的BYE请求无法到达目的地。当原始BYE请求格式错误且其他方忽略它时，也可能出现这种情况。

您是否尝试使用SIPp在本地调试服务器？您还可以运行Ethereal（Wireshark）来检查您的流量。

Answer 2

“via_cnt == 1092”也非常可疑。

顺便说一句，您似乎正在构建B2BUA，因为您甚至在向本地电话发送邀请之前接受来自外部的呼叫（1234）。如果本地电话接受不同的参数，接受不同的编解码器等，你会被搞砸，因为你告诉本地电话将媒体直接发送给原来的来电者。他们真的应该将他们的媒体发送到您的服务器，这将传递（或者如果需要转码）。

如果您不想这样做（即您不想充当媒体中继和可能的代码转换器），您需要将INVITE转发到本地电话，然后转发任何响应等。基本上采取更多措施作为SIP代理服务器而不是SIP B2BUA。

Answer 3

我建议检查调用腿是否接受BYE请求（看起来它接受但是......）以及它如何处理此请求。你真正需要的是类似于174.37.45.134的日志。似乎问题落后于.134（超时由.134生成）。

首先，我看到你打破几个基本的呼叫处理规则，这可能会让你遇到麻烦： - 您缺少尝试响应原始呼叫支路。如果始发者的SIP堆栈确实等待这一点，则可能导致呼叫ID未被真正记录。是的，这是有缺陷的行为，但我们生活在现实世界中。标准说要用Trying ASAP响应（甚至在你进行路由之前，就在呼叫认证之后） - 您甚至在向被叫方发起传出邀请之前，正在与主叫方完全建立呼叫会话。这是错误的逻辑。至少因为在拨打电话失败的情况下，发票人将被收费。

如果您可以快速完成此操作，我建议您先修复呼叫设置顺序。你无论如何都需要这个，并且有可能这将解决呼叫终止：

INVITE  ->
TRYING  <-
        -> INVITE
        <- TRYING
        <- RINGING
RINGING <-
        <- OK
OK      <-
ACK     ->
        -> ACK

Answer 4

如果您希望符合RFC 3261，则强制要求您发送的“Via”标头包含可选（！）“branch”参数。

参见RFC3261 ss 20.42：

即使此规范要求分支参数为在所有请求中，标题字段的BNF表示它是可选的。这允许与RFC 2543元素互操作，它不必插入分支参数。

Answer 5

RFC 3261规定To和From字段中关联的数字（URI）保持不变。如果涉及NAT'ing，IP可以改变，但数字必须保持不变。如果您注意到，BYE标题中的“收件人”和“发件人”字段将被交换，从而使其成为格式错误的数据包。

SIP BYE消息的问题

5 个答案: