我正在将WebRTC本机实现到Android中。我能够按照此处http://www.webrtc.org/native-code/android的说明编译和运行代码,但是遇到了一个问题,其中apprtc.appspot.com显然未按预期返回通道 token :

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

当在这里https://code.google.com/p/webrtc/source/browse/trunk/talk/examples/android/src/org/appspot/apprtc/AppRTCClient.java?r=5847检查AppRTCClient.java的源代码时,我在第234ff行发现了一个有趣的注释,正是该应用失败的功能:
// 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代码解析,并且似乎没有从服务器返回的响应html中正确地解析了通道 token (难怪)。

我没有进一步调查通道 token 是否存在和/或是否正确。相反,我怀疑代码(trunk!)可能已经过时,被谷歌搜索并在Github上找到了这个项目:https://github.com/pristineio/webrtc-android

相同的函数(在232ff行)将&t = json作为参数传递,这可能是因为它希望服务器返回Json。然后,该函数在解析Json时失败,请猜测原因。答对了!因为返回的是相同的html页面,带有或不带有参数。
// 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。

因此,似乎Google的Fischman先生在此同时更新了代码,但webrtc上的代码不是最新的,Github上的pristineio/webrtc-android似乎知道曾经使apprtc.appspot.com返回Json而不是Json的参数。 html,但不再这样做。

我用谷歌搜索,但找不到apprtc.appspot.com的服务器代码,但我记得以前看过它(我相信这是一个Python项目)。
  • 是否有人链接到该源代码?

  • 然后,我搜索了该URL的参数,然后找到列出了这些参数的两个页面,其中包括http://samdutton.github.io/webrtc/samples/web/content/apprtc/params.html,但是没有一个页面显示参数或替代URL以Json的形式请求结果。
  • 有人知道正确的参数(或网址)是什么吗?
  • 还是有人知道完整规格吗?
  • 还是没有人链接到AppRTCClient.Java的最新有效源代码?

  • 谢谢!

    最佳答案

    遇到与您相同的问题。我在这里找到了代码:

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

    当前正在查看更改,以查看可能会破坏它的内容。如果发现有问题,请及时更新此答案。

    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 Channel API迁移到Websocket通道。

    08-17 13:26