我是MjSip的新手,我使用MjUa来创建客户机。我想连接到一个星号服务器。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}
但是我仍然有488个错误,请帮帮我。如何更改"MjUa“配置文件?
以下是所有消息日志:
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-----
发布于 2014-03-28 19:05:48
有点晚,但很多时候,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)是否至少有一个代码可以协商。
从张贴的日志中:
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
G711似乎是被请求的,但在服务器端不可用。因此,服务器拒绝RTP通道。
发布于 2013-06-21 14:08:30
我在使用Snom 300手机联系星号服务器时也犯了同样的错误。在电话上关闭RTP加密对我有效。
在V7固件上,这是在"V7:标识- RTP设置(节):RTP加密“中。显然,在V7上,RTP加密默认为:srtp。
我不知道根本原因是否是星号服务器配置错误(我没有运行它),但至少这解决了问题。
发布于 2020-01-07 01:46:09
对我来说,这是我的VOIP提供商的服务器端设置,只期望加密连接。在我恢复到客户机中的明文连接之后,我忘记了这一点。
https://stackoverflow.com/questions/15852013
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