从Web客户端呼叫到软电话Twinkle已收到，但在下一刻断开连接

时间：2014-05-11 11:23:48

标签： sip webrtc freeswitch sip-server sipml

我正在尝试从网络客户端SipML5实时演示页面调用freeswitch的注册用户。

现在有两个问题。

1。有时用户1002已成功连接，并且能够在闪烁时拨打用户1001。 但一旦收到已断开连接，就会

。

此日志粘贴在此处。请看看。

FreeSwitch的日志 

2014-05-11 00:02:47.548348 [CONSOLE] mod_voicemail.c:4066 Event Thread Started
2014-05-11 00:06:01.488350 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/1002@192.168.62.6 [3613d285-aa9d-4808-b5b6-85b60ab61a03]
2014-05-11 00:06:01.588349 [INFO] mod_dialplan_xml.c:558 Processing 1002 <1002>->1001 in context default
2014-05-11 00:06:01.588349 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *1 execute_extension::dx XML features
2014-05-11 00:06:01.588349 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1002.2014-05-11-00-06-01.wav
2014-05-11 00:06:01.588349 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *3 execute_extension::cf XML features
2014-05-11 00:06:01.588349 [INFO] switch_ivr_async.c:3631 Bound B-Leg: *4 execute_extension::att_xfer XML features
2014-05-11 00:06:01.588349 [NOTICE] switch_channel.c:1054 New Channel sofia/internal/sip:1001@192.168.62.6:5075 [e2a5e50b-87fd-41a6-bd52-c59aa1f7abf2]
2014-05-11 00:06:01.588349 [NOTICE] sofia.c:6287 Ring-Ready sofia/internal/sip:1001@192.168.62.6:5075!
2014-05-11 00:06:01.608350 [INFO] switch_ivr_originate.c:1191 Sending early media
2014-05-11 00:06:01.608350 [WARNING] switch_core_media.c:3455 Crypto not negotiated but required.
2014-05-11 00:06:01.608350 [ERR] mod_sofia.c:2201 CODEC NEGOTIATION ERROR.  SDP:
v=0
o=doubango 1983 678901 IN IP4 192.168.62.6
s=-
c=IN IP4 192.168.62.6
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 51232 RTP/AVP 111 8 0 101
c=IN IP4 192.168.62.6
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000; sprop-maxcapturerate=48000; stereo=0; sprop-stereo=0; useinbandfec=0; usedtx=0
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ZXnus9mDUN0KeIAH4oFY08wP7RU2jO0QCz0ATrrx
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:QVlx1ZwghQbqGNN1vldBCSx2+xNH1IOhoDYFuc85
a=pcfg:1 t=1 a=1,2
a=pcfg:2 t=2 a=1,2
a=pcfg:3 t=3
a=rtcp-mux
a=ssrc:2867390165 cname:019f5bad1e89245827816436cc655e35
a=ssrc:2867390165 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:2867390165 label:doubango@audio
a=ice-ufrag:KORey5kawUsPr7u
a=ice-pwd:iXgoMD0A1kh0dPEU9KNXPl
a=candidate:kMxPxHHUa 1 udp 2130706431 192.168.62.6 51232 typ host
a=candidate:kMxPxHHUa 2 udp 2130706430 192.168.62.6 51233 typ host
a=candidate:DeKKSOIA3 1 udp 2130706175 192.168.49.170 53856 typ host
a=candidate:DeKKSOIA3 2 udp 2130706174 192.168.49.170 53857 typ host

2014-05-11 00:06:01.608350 [NOTICE] switch_channel.c:3432 Hangup sofia/internal/1002@192.168.62.6 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
2014-05-11 00:06:01.608350 [NOTICE] switch_ivr_originate.c:3782 Hangup sofia/internal/sip:1001@192.168.62.6:5075 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2014-05-11 00:06:01.608350 [NOTICE] switch_ivr_originate.c:2707 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL]
2014-05-11 00:06:01.608350 [INFO] mod_dptools.c:3234 Originate Failed.  Cause: ORIGINATOR_CANCEL
2014-05-11 00:06:01.608350 [NOTICE] switch_core_session.c:1622 Session 2 (sofia/internal/sip:1001@192.168.62.6:5075) Ended
2014-05-11 00:06:01.608350 [NOTICE] switch_core_session.c:1626 Close Channel sofia/internal/sip:1001@192.168.62.6:5075 [CS_DESTROY]
2014-05-11 00:06:01.608350 [NOTICE] switch_core_session.c:1622 Session 1 (sofia/internal/1002@192.168.62.6) Ended
2014-05-11 00:06:01.608350 [NOTICE] switch_core_session.c:1626 Close Channel sofia/internal/1002@192.168.62.6 [CS_DESTROY]

WebRTC2Sip的日志 

MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()" 
file: "src/tnet_socket.c" 
line: "143" 
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()" 
file: "src/tnet_socket.c" 
line: "143" 
MSG: (SYSTEM)NETWORK ERROR ==>Success
***ERROR: function: "tnet_getaddrinfo()" 
file: "src/tnet_utils.c" 
line: "928" 
MSG: getaddrinfo(family=10, node=:: and service=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()" 
file: "src/tnet_socket.c" 
line: "143" 
MSG: tnet_getaddrinfo(family=10, hostname=:: and port=0) failed: [Name or service not known]
***ERROR: function: "tnet_socket_create_2()" 
file: "src/tnet_socket.c" 
line: "143" 
MSG: (SYSTEM)NETWORK ERROR ==>Success

和

Log of Chrome Java Script

2。有时打电话根本不打。网络客户端连接只停留在州 - ＆＃34; 正在进行呼叫... ＆＃34;按下通话按钮后。为此，FreeSwitch和webrtc2sip中不会生成任何日志。的 Chrome JS log is here.

webrtc2sip config.xml文件内容是这个 - ＆gt; 

<?xml version="1.0" encoding="utf-8" ?>
<!-- Please check the technical guide (http://webrtc2sip.org/technical-guide-1.0.pdf) for more information on how to adjust this file -->
<config>

  <debug-level>ERROR</debug-level>

  <transport>udp;*;10060</transport>
  <transport>ws;*;10060</transport>
  <transport>wss;*;10062</transport>
  <!--transport>tcp;*;10063</transport-->
  <!--transport>tls;*;10064</transport-->

  <enable-rtp-symetric>yes</enable-rtp-symetric>
  <enable-100rel>yes</enable-100rel>
  <enable-media-coder>yes</enable-media-coder>
  <enable-videojb>no</enable-videojb>
  <!--video-size-pref>vga</video-size-pref-->
  <rtp-buffsize>65535</rtp-buffsize>
  <avpf-tail-length>100;400</avpf-tail-length>
  <srtp-mode>optional</srtp-mode>
  <srtp-type>sdes;dtls</srtp-type>
  <dtmf-type>rfc4733</dtmf-type>

  <codecs>pcma;pcmu;speex;ilbc;opus</codecs>
  <codec-opus-maxrates>48000;48000</codec-opus-maxrates>

  <stun-server>stun.l.google.com;19302;stun-user@doubango.org;stun-password</stun-server>
  <enable-icestun>yes</enable-icestun>

  <max-fds>-1</max-fds>

  <!--nameserver>66.66.66.6</nameserver-->

  <!--ssl-certificates>
    C:/Projects/ssl/priv.pem;
    C:/Projects/ssl/pub.pem;
    C:/Projects/ssl/ca-cert.pem;
  </ssl-certificates-->

  <!-- ***CLICK-TO-CALL SERVICE*** -->

  <transport>c2c;*;10070</transport>
  <transport>c2cs;*;10072</transport>
  <database>sqlite;*</database>
  <!--account-mail>smtps;*;*;auth.smtp.1and1.fr;465;noreply@example.com;noreply@example.com;mysecret</account-mail-->
  <!--account-sip-caller>*;sip:a@example.com;a;example.com;mysecret</account-sip-caller-->

</config>

FreeSwitch版本 - ＆gt; 1.5.12b

WebRTC2Sip版本 - ＆gt; 2.6.0

Chrome版本 - ＆gt; 31.0.1650.63

OS - Ubuntu 12.04

如果需要更多信息，请与我们联系。

由于

1 个答案:

答案 0 :(得分：1)

可能原因在于：

2014-05-11 00:06:01.608350 [WARNING] switch_core_media.c:3455 Crypto not negotiated but required.
2014-05-11 00:06:01.608350 [ERR] mod_sofia.c:2201 CODEC NEGOTIATION ERROR.  SDP:

后来导致这个：

2014-05-11 00:06:01.608350 [NOTICE] switch_channel.c:3432 Hangup sofia/internal/1002@192.168.62.6 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

看起来有编解码器错误

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