5

我是 MjSip 的新手，我使用 MjUa 来创建客户端。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程序。我使用这个配置：

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

但我仍然收到 488 错误，请帮助我。如何更改“MjUa”配置文件？

这是所有消息日志：

INVITE sip:57@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK2bfdff77
Max-Forwards: 70
To: "Alice" <sip:57@192.168.0.254:5060>
From: "aziz" <sip:157@192.168.0.254>;tag=350164683297
Call-ID: 728007708208@192.168.0.57
CSeq: 1 INVITE
Contact: <sip:157@192.168.0.57>
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:157@192.168.0.254>;tag=350164683297
To: "Alice" <sip:57@192.168.0.254:5060>;tag=as3f160681
Call-ID: 728007708208@192.168.0.57
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:57@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK2bfdff77
Max-Forwards: 70
To: "Alice" <sip:57@192.168.0.254:5060>;tag=as3f160681
From: "aziz" <sip:157@192.168.0.254>;tag=350164683297
Call-ID: 728007708208@192.168.0.57
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:57@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK644461b7
Max-Forwards: 70
To: "Alice" <sip:57@192.168.0.254:5060>
From: "aziz" <sip:157@192.168.0.254>;tag=350164683297
Call-ID: 728007708208@192.168.0.57
CSeq: 2 INVITE
Contact: <sip:157@192.168.0.57>
Expires: 3600
User-Agent: mjsip 1.7
Authorization: Digest username="157", realm="asterisk", nonce="6e640e9a", uri="sip:57@192.168.0.254:5060", 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:157@192.168.0.254>;tag=350164683297
To: "Alice" <sip:57@192.168.0.254:5060>;tag=as3f160681
Call-ID: 728007708208@192.168.0.57
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:57@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.57:5060;rport;branch=z9hG4bK644461b7
Max-Forwards: 70
To: "Alice" <sip:57@192.168.0.254:5060>;tag=as3f160681
From: "aziz" <sip:157@192.168.0.254>;tag=350164683297
Call-ID: 728007708208@192.168.0.57
CSeq: 2 ACK
User-Agent: mjsip 1.7
Content-Length: 0

-----End-of-message-----
4

4 回答 4

5

有点晚了，但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

似乎请求了 G711，但在服务器端不可用。因此，服务器拒绝 RTP 通道。

于 2014-03-28T19:05:48.013 回答
3

我在使用 Snom 300 电话联系 Asterisk 服务器时遇到了同样的错误。在手机上关闭 RTP 加密对我有用。

在 V7 固件上，它位于：“V7：身份 - RTP 设置（部分）：RTP 加密”。显然，在 V7 上，RTP 加密默认开启：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），但至少这解决了这个问题。

于 2013-06-21T14:08:30.250 回答
2

对我来说，这是我的 VOIP 提供商的服务器端设置，只需要加密连接。在客户端恢复为纯文本连接后，我忘记了它。

于 2020-01-07T01:46:09.820 回答
1

我在 Zoiper5 桌面应用程序中遇到了这个错误。该问题可能通过设置RTCP Feedback-> OFF解决，之前我使用“兼容模式”，因此它是 488 错误的最可能原因。另外，我已将编解码器的顺序更改为：G.711 mulaw；一条法律; GSM FR；G.722 而将 OPUS 编解码器移动到最不优选的现场编解码器的顺序。

于 2021-08-16T11:27:01.423 回答