AppRTC服务器返回html而不是Json

时间:2015-01-06 04:46:13

标签: android webrtc apprtcdemo

我正在将WebRTC本地实现到Android中。我能够编译并运行这里描述的代码http://www.webrtc.org/native-code/android,但我遇到了一个问题,其中apprtc.appspot.com显然没有按照假设返回通道令牌:

01-05 20:01:51.230  15488-15488/org.appspot.apprtc E/AppRTCDemoActivity﹕ Fatal error: Missing channelToken in HTML: <!DOCTYPE html>
    <!--
    *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
    *
    *  Use of this source code is governed by a BSD-style license
    *  that can be found in the LICENSE file in the root of the source
    *  tree.
    -->
    <html>
    <head>
    <title>WebRTC Reference App</title>
    <meta charset="utf-8">
    <meta name="description" content="WebRTC reference app">
    <meta name="viewport" content="width=device-width, user-scalable=no, initial-scale=1, maximum-scale=1">
    <script type='text/javascript'>window.mod_pagespeed_start = Number(new Date());</script><link rel="canonical" href="https://apprtc.appspot.com/room/42272483?r=fh">
    <link rel="stylesheet" href="https://1-ps.googleusercontent.com/sxk/j1xSZgZ8VRf8QgCJPihVzk5mBR/s.apprtc.appspot.com/apprtc.appspot.com/css/A.main.css.pagespeed.cf.mluzaRyZGPUUQu3CIFwW.css">
    </head>
    <body><noscript><meta HTTP-EQUIV="refresh" content="0;url='https://apprtc.appspot.com/r/42272483?r=fh&amp;PageSpeed=noscript'" /><style><!--table,div,span,font,p{display:none} --></style><div style="display:block">Please click <a href="https://apprtc.appspot.com/r/42272483?r=fh&amp;PageSpeed=noscript">here</a> if you are not redirected within a few seconds.</div></noscript>
    <div id="videos">
    <video id="mini-video" autoplay muted></video>
    <canvas id="remote-canvas"></canvas>
    <video id="remote-video" autoplay></video>
    <video id="local-video" autoplay muted></video>
    </div>
    <footer>
    <div id="sharing">
    <div id="room-link">Waiting for someone to join this room: <a href="https://apprtc.appspot.com/room/42272483?r=fh" target="_blank">https://apprtc.appspot.com/room/42272483?r=fh</a></div>
    </div>
    <div id="info"></div>
    <div id="status"></div>
    </footer>
    <script src="https://1-ps.googleusercontent.com/sxk/j1xSZgZ8VRf8QgCJPihVzk5mBR/s.apprtc.appspot.com/apprtc.appspot.com/js/stats.js.pagespeed.jm.A-w15PL7V0sRNC026ANH.js"></script>
    <script src="https://1-ps.googleusercontent.com/sxk/j1xSZgZ8VRf8QgCJPihVzk5mBR/s.apprtc.appspot.com/apprtc.appspot.com/js/signaling.js.pagespeed.jm.QZ7VRUXKfNhiyV7jHkmW.js"></script>
    <script src="https://1-ps.googleusercontent.com/sxk/j1xSZgZ8VRf8QgCJPihVzk5mBR/s.apprtc.appspot.com/apprtc.appspot.com/js/infobox.js.pagespeed.jm.C9t_78UyYtO6bMRljt_L.js"></script>
    <script src="https://1-ps.googleusercontent.com/sxk/j1xSZgZ8VRf8QgCJPihVzk5mBR/s.apprtc.appspot.com/apprtc.appspot.com/js/sdputils.js.pagespeed.jm.DWy54ENwSOTLQKw10p1o.js"></script>
    <script src="https://1-ps.googleusercontent.com/sxk/j1xSZgZ8VRf8QgCJPihVzk5mBR/s.apprtc.appspot.com/apprtc.appspot.com/js/util.js.pagespeed.jm.q3iuB_S1TC2eBJC_RFlb.js"></script>
    <script src="https://1-ps.googleusercontent.com/sxk/j1xSZgZ8VRf8QgCJPihVzk5mBR/s.apprtc.appspot.com/apprtc.appspot.com/js/main.js.pagespeed.jm.mR4sH1O_ReNLDaNiS3o_.js"></script>
    <script src="https://1-ps.googleusercontent.com/sxk/j1xSZgZ8VRf8QgCJPihVzk5mBR/s.apprtc.appspot.com/apprtc.appspot.com/js/adapter.js.pagespeed.jm.-Ip1bBjviqtsxeGluAGS.js"></script>
    <script type="text/javascript">var params={errorMessages:[],isLoopback:false,roomId:'42272483',roomLink:'https://apprtc.appspot.com/room/42272483?r=fh',mediaConstraints:{"audio":true,"video":true},offerConstraints:{"optional":[],"mandatory":{}},peerConnectionConfig:{"iceServers":[]},peerConnectionConstraints:{"optional":[{"googImprovedWifiBwe":true}]},turnRequestUrl:'https://computeengineondemand.appspot.com/turn?username=280585048&key=4080218913',turnTransports:'',audioSendBitrate:'',audioSendCodec:'',audioRecvBitrate:'',audioRecvCodec:'',isStereoscopic:'',opusMaxPbr:'',opusFec:'',opusStereo:'',videoSendBitrate:'',videoSendInitialBitrate:'',videoSendCodec:'',videoRecvBitrate:'',videoRecvCodec:'',wssUrl:'wss://apprtc-ws.webrtc.org:443/ws',wssPostUrl:'https://apprtc-ws.webrtc.org:443'};initialize();</script>
    <script>(function(i,s,o,g,r,a,m){i['GoogleAnalyticsObject']=r;i[r]=i[r]||function(){(i[r].q=i[r].q||[]).push(arguments)},i[r].l=1*new Date();a=s.cr

在这里检查AppRTCClient.java的源代码https://code.google.com/p/webrtc/source/browse/trunk/talk/examples/android/src/org/appspot/apprtc/AppRTCClient.java?r=5847时,我在第234ff行发现了一个有趣的评论,正是应用程序失败的功能:

// Fetches |url| and fishes the signaling parameters out of the HTML via
// regular expressions.
//
// TODO(fischman): replace this hackery with a dedicated JSON-serving URL in
// apprtc so that this isn't necessary (here and in other future apps that
// want to interop with apprtc).
private AppRTCSignalingParameters getParametersForRoomUrl(String url)
        throws IOException {

    // ...

}

在函数中发生了很多html代码的解析,似乎没有从服务器返回的响应html中正确解析通道令牌(难怪)。

我没有进一步调查通道令牌是否存在和/或是否正确。相反,我怀疑代码(trunk!)可能已过时,用Google搜索并在Github上找到此项目:https://github.com/pristineio/webrtc-android

同样的函数,在这里的第23222行,传入&amp; t = json作为参数,因为它期望服务器返回Json。解析Json时函数失败,猜猜原因。答对了!因为返回相同的html页面,包含或不包含参数。

// Fetches |url| and fishes the signaling parameters out of the JSON.
private AppRTCSignalingParameters getParametersForRoomUrl(String url)
        throws IOException, JSONException {
    url = url + "&t=json";

    // ...

}

为了完整起见,失败的完整URL:

https://apprtc.appspot.com/?r=00000000&t=json

其中00000000是房间号。

所以看来谷歌的Fischman先生在此期间更新了代码,但是webrtc的代码不是最新的,而且Github上的pristineio / webrtc-android似乎知道曾经让apprtc.appspot.com返回的参数Json而不是html,但它不再这样做了。

我用Google搜索但是找不到apprtc.appspot.com的服务器代码,但我记得以前见过它(我相信这是一个Python项目)。

  • 是否有人拥有该源代码的链接?

然后我搜索了该URL的参数,我找到了列出这些URL的两个页面,包括http://samdutton.github.io/webrtc/samples/web/content/apprtc/params.html,但没有一个显示参数或替代URL来请求结果为Json。

  • 有人知道正确的参数(或网址)是什么吗?
  • 或者有人知道全部规格是什么吗?
  • 或者是否有人链接到AppRTCClient.Java的最新和有效的源代码?

谢谢!

1 个答案:

答案 0 :(得分:3)

遇到和你一样的问题。我在这里找到了代码:

https://github.com/GoogleChrome/webrtc/tree/master/samples/web/content/apprtc

目前正在查看更改以查看可能会破坏它的内容。如果我找到了什么,我会更新这个答案。

更新:看起来他们在此提交中做了一些重大更改:

https://github.com/GoogleChrome/webrtc/commit/c36b88475fab8a3e4436a87a7ea84265b0e13a8a#diff-c3e41e94913c93dfe31babd4830c3065

他们正在从GAE频道api转移到websocket频道。