webrtc音频限制带宽

我们使用webrtc的时候遇到过带宽占用太高的问题。查看别人的博客也发现他们的项目一般都是支持一个服务器5到6人的语音。这是成本太高的问题。

其实webrtc自己是可以设置这些的。

仔细阅读webrtc的源码，我设置如下的参数，使得带宽减少了一半多。一个5M服务器可以支持近30人。

private static final boolean _preferIsac = true;
private static final int _audioStartBitrate = 0;

调用下面的两个函数：

private static String preferCodec(
            String sdpDescription, String codec, boolean isAudio) {
        String[] lines = sdpDescription.split("\r\n");
        int mLineIndex = -1;
        String codecRtpMap = null;
        // a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>]
        String regex = "^a=rtpmap:(\\d+) " + codec + "(/\\d+)+[\r]?$";
        Pattern codecPattern = Pattern.compile(regex);
        String mediaDescription = "m=video ";
        if (isAudio) {
            mediaDescription = "m=audio ";
        }
        for (int i = 0; (i < lines.length)
                && (mLineIndex == -1 || codecRtpMap == null); i++) {
            if (lines[i].startsWith(mediaDescription)) {
                mLineIndex = i;
                continue;
            }
            Matcher codecMatcher = codecPattern.matcher(lines[i]);
            if (codecMatcher.matches()) {
                codecRtpMap = codecMatcher.group(1);
            }
        }
        if (mLineIndex == -1) {
            Log.w(TAG, "No " + mediaDescription + " line, so can't prefer " + codec);
            return sdpDescription;
        }
        if (codecRtpMap == null) {
            Log.w(TAG, "No rtpmap for " + codec);
            return sdpDescription;
        }
        Log.d(TAG, "Found " +  codec + " rtpmap " + codecRtpMap + ", prefer at "
                + lines[mLineIndex]);
        String[] origMLineParts = lines[mLineIndex].split(" ");
        if (origMLineParts.length > 3) {
            StringBuilder newMLine = new StringBuilder();
            int origPartIndex = 0;
            // Format is: m=<media> <port> <proto> <fmt> ...
            newMLine.append(origMLineParts[origPartIndex++]).append(" ");
            newMLine.append(origMLineParts[origPartIndex++]).append(" ");
            newMLine.append(origMLineParts[origPartIndex++]).append(" ");
            newMLine.append(codecRtpMap);
            for (; origPartIndex < origMLineParts.length; origPartIndex++) {
                if (!origMLineParts[origPartIndex].equals(codecRtpMap)) {
                    newMLine.append(" ").append(origMLineParts[origPartIndex]);
                }
            }
            lines[mLineIndex] = newMLine.toString();
            Log.d(TAG, "Change media description: " + lines[mLineIndex]);
        } else {
            Log.e(TAG, "Wrong SDP media description format: " + lines[mLineIndex]);
        }
        StringBuilder newSdpDescription = new StringBuilder();
        for (String line : lines) {
            newSdpDescription.append(line).append("\r\n");
        }
        return newSdpDescription.toString();
    }

 private static String setStartBitrate(String codec, boolean isVideoCodec,
                                          String sdpDescription, int bitrateKbps) {
        String[] lines = sdpDescription.split("\r\n");
        int rtpmapLineIndex = -1;
        boolean sdpFormatUpdated = false;
        String codecRtpMap = null;
        // Search for codec rtpmap in format
        // a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>]
        String regex = "^a=rtpmap:(\\d+) " + codec + "(/\\d+)+[\r]?$";
        Pattern codecPattern = Pattern.compile(regex);
        for (int i = 0; i < lines.length; i++) {
            Matcher codecMatcher = codecPattern.matcher(lines[i]);
            if (codecMatcher.matches()) {
                codecRtpMap = codecMatcher.group(1);
                rtpmapLineIndex = i;
                break;
            }
        }
        if (codecRtpMap == null) {
            Log.w(TAG, "No rtpmap for " + codec + " codec");
            return sdpDescription;
        }
        Log.d(TAG, "Found " +  codec + " rtpmap " + codecRtpMap
                + " at " + lines[rtpmapLineIndex]);

        // Check if a=fmtp string already exist in remote SDP for this codec and
        // update it with new bitrate parameter.
        regex = "^a=fmtp:" + codecRtpMap + " \\w+=\\d+.*[\r]?$";
        codecPattern = Pattern.compile(regex);
        for (int i = 0; i < lines.length; i++) {
            Matcher codecMatcher = codecPattern.matcher(lines[i]);
            if (codecMatcher.matches()) {
                Log.d(TAG, "Found " +  codec + " " + lines[i]);
                if (isVideoCodec) {
                    lines[i] += "; " + VIDEO_CODEC_PARAM_START_BITRATE
                            + "=" + bitrateKbps;
                } else {
                    lines[i] += "; " + AUDIO_CODEC_PARAM_BITRATE
                            + "=" + (bitrateKbps * 1000);
                }
                Log.d(TAG, "Update remote SDP line: " + lines[i]);
                sdpFormatUpdated = true;
                break;
            }
        }

        StringBuilder newSdpDescription = new StringBuilder();
        for (int i = 0; i < lines.length; i++) {
            newSdpDescription.append(lines[i]).append("\r\n");
            // Append new a=fmtp line if no such line exist for a codec.
            if (!sdpFormatUpdated && i == rtpmapLineIndex) {
                String bitrateSet;
                if (isVideoCodec) {
                    bitrateSet = "a=fmtp:" + codecRtpMap + " "
                            + VIDEO_CODEC_PARAM_START_BITRATE + "=" + bitrateKbps;
                } else {
                    bitrateSet = "a=fmtp:" + codecRtpMap + " "
                            + AUDIO_CODEC_PARAM_BITRATE + "=" + (bitrateKbps * 1000);
                }
                Log.d(TAG, "Add remote SDP line: " + bitrateSet);
                newSdpDescription.append(bitrateSet).append("\r\n");
            }

        }
        return newSdpDescription.toString();
    }

这两个函数的使用方法可以在webrtc的源码中获得，webrtc-master-webrtc\examples\androidapp\src\org\appspot\apprtc\PeerConnectionClient.java

大家也可以尝试其他的方法。或许能够得到更喜人的方案。我们的结果是ICE连接占用23KB左右，声音相关35KB。

如果有更好的方案请邮件发我：

fatestaymq2@foxmail.com