““ SIP / 2.0 488此处不可接受”错误

Question

我是MjSip的新手，我使用MjUa创建客户端。我想连接到星号服务器。它支持G.711，但我无法配置我的应用。我使用此配置：

 media=audio 4000 rtp/avp {audio 0 PCMU 8000 160, audio 8 PCMA 8000 160}

但是我仍然收到488错误请帮我。如何更改“ MjUa”配置文件？

---

这里是所有消息日志：

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK2bfdff77
Max-Forwards: 70
To: "Alice" <sip:[email protected]:5060>
From: "aziz" <sip:[email protected]>;tag=350164683297
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Expires: 3600
User-Agent: mjsip 1.7
Content-Length: 141
Content-Type: application/sdp

v=0
o=157 0 0 IN IP4 192.168.0.57
s=-
c=IN IP4 192.168.0.57
t=0 0
m=audio 4000 rtp/avp 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
-----End-of-message-----

1365314026097: 10:23:46.097 Sun 07 Apr 2013, 192.168.0.254:5060/udp (519 bytes) received
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.57:5060;branch=z9hG4bK2bfdff77;received=192.168.0.57;rport=5060
From: "aziz" <sip:[email protected]>;tag=350164683297
To: "Alice" <sip:[email protected]:5060>;tag=as3f160681
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6e640e9a"
Content-Length: 0

-----End-of-message-----

1365314026107: 10:23:46.107 Sun 07 Apr 2013, 192.168.0.254:5060/udp (326 bytes) sent
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK2bfdff77
Max-Forwards: 70
To: "Alice" <sip:[email protected]:5060>;tag=as3f160681
From: "aziz" <sip:[email protected]>;tag=350164683297
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: mjsip 1.7
Content-Length: 0

-----End-of-message-----

1365314026151: 10:23:46.151 Sun 07 Apr 2013, 192.168.0.254:5060/udp (706 bytes) sent
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK644461b7
Max-Forwards: 70
To: "Alice" <sip:[email protected]:5060>
From: "aziz" <sip:[email protected]>;tag=350164683297
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>
Expires: 3600
User-Agent: mjsip 1.7
Authorization: Digest username="157", realm="asterisk", nonce="6e640e9a", uri="sip:[email protected]:5060", algorithm=MD5, response="84ff5e12b8325a153e09ac2e316f5b1f"
Content-Length: 141
Content-Type: application/sdp

v=0
o=157 0 0 IN IP4 192.168.0.57
s=-
c=IN IP4 192.168.0.57
t=0 0
m=audio 4000 rtp/avp 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
-----End-of-message-----

1365314026152: 10:23:46.152 Sun 07 Apr 2013, 192.168.0.254:5060/udp (450 bytes) received
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.0.57:5060;branch=z9hG4bK644461b7;received=192.168.0.57;rport=5060
From: "aziz" <sip:[email protected]>;tag=350164683297
To: "Alice" <sip:[email protected]:5060>;tag=as3f160681
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

-----End-of-message-----

1365314026155: 10:23:46.155 Sun 07 Apr 2013, 192.168.0.254:5060/udp (326 bytes) sent
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK644461b7
Max-Forwards: 70
To: "Alice" <sip:[email protected]:5060>;tag=as3f160681
From: "aziz" <sip:[email protected]>;tag=350164683297
Call-ID: [email protected]
CSeq: 2 ACK
User-Agent: mjsip 1.7
Content-Length: 0

-----End-of-message-----

Answer 1

有点晚了，但是通常这与编解码器不兼容有关。对于遇到此问题的任何人，他们应该检查双方（服务器和客户端）是否至少有一个可以协商的代码。

从发布的日志中：

a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

似乎已请求G711，但在服务器端不可用。因此，服务器拒绝RTP通道。

Answer 2

我使用Snom 300电话联系Asterisk服务器时遇到相同的错误。关闭手机上的RTP加密对我有用。

在V7固件上，它位于：“ V7：标识-RTP设置（部分）：RTP加密”。显然，在V7上，RTP加密默认情况下处于打开状态：http://wiki.snom.com/wiki/index.php/Settings/user_srtp

我不知道根本原因是否是Asterisk服务器配置错误（我没有运行它），但至少可以解决此问题。

Answer 3

对我来说，这是我的VOIP提供程序的服务器端设置，期望仅加密连接。在客户端恢复为纯文本连接后，我忘记了它。