webrtc - 视频获得blob，但它仍然是黑色的

问题描述

我使用chrome 21运行我的webrtc代码。

I run my webrtc code with chrome 21.

如果我在同一个chrome中打开两个标签，然后打开包含webrtc代码的页面。
一个标签用于发送视频流;一个标签用于接收视频流
工作正常。

If I open two tabs in the same chrome, and then open page with webrtc code inside.One tab is for sending video stream; one tab is for receiving video streamIt works fine.

但是，如果我打开两个隐身模式或两个不同的Chrome浏览器页面，
我可以正确获取 sdp 和候选人信息。看来视频可以解码信息。

However, If I open page with two Incognito mode or two different chrome browser,I can get the sdp and candidate information correctly. It seems that video can decode the information.

在远程视频中，我只能看到

In remote video, I can see only

此外，它似乎崩溃了。我试图点击关闭铬但没用。

Besides, it seems crash. I tried to click "close chrome" but useless.

有没有人有类似的问题？

Does anyone have any similar problem?

推荐答案

测试时 WebRTC ，我发现当我们在错误的位置调用 peerConnection.addStream（...）时会出现这种情况--- -

While testing WebRTC, I found that such condition occurs when we call peerConnection.addStream(…) in the wrong place ----

你必须记住，在 WebRTC 中订购非常重要！

You must remember that ordering highly matters in WebRTC!

空白视频在以下情况下发生：

Blank video occurs in following cases:

1. 当您的SSL证书已过期或条目无效时，您正在使用STUN。


2. 您正在使用STUN，但它是公司防火墙，或医院网络或专用网络，它阻止或隐藏外部IP地址或某些端口。


3. 两个对等方都无效sendrecv / sendonly / recvonly对


4. Offerer没有附加流，或者Firefox是在用户附加的情况下失败的情况ly音频流然而你正在使用 OfferToReceiveVideo：true


5. 你正在检查 HTMLMediaElement.HAVE_CURRENT_DATA 或 mediaElement.paused 或 mediaElement.currentTime 虽然它是android，已知有关于这些的问题物业。

1. You're using STUN whilst your SSL certificate is either expired or it has invalid entries.
2. You're using STUN however it is corporate firewall, or hospital network or private network which is blocking or hiding external ip addresses or some ports.
3. Both peers has invalid sendrecv/sendonly/recvonly pairs
4. Offerer didn't attach the stream or it is Firefox which fails in cases when user attached only audio stream however you're using OfferToReceiveVideo:true
5. You're checking for HTMLMediaElement.HAVE_CURRENT_DATA or mediaElement.paused or mediaElement.currentTime whilst it is android which has known issues regarding these properties.

解决方案？

Solutions?

1. 使用从XirSys转发或。


2. 确保您使用的是有效的SSL证书或使用HTTP。


3. 确保提供者附加了流;还要确保根据附加的流使用 OfferToReceiveAudio / OfferToReceiveVideo 。


4. 确保您没有修改SDP;也尝试比较两个同行之间的SDP和查找不匹配。

1. Use TURN from XirSys or install your own.
2. Make sure that you're using valid SSL certificate or use HTTP instead.
3. Make sure that offerer attached the stream; also make sure that OfferToReceiveAudio/OfferToReceiveVideo are used according to stream(s) attached.
4. Make sure that you didn't modify the SDP; also try to compare SDP between both peers and find-out mismatches.

代码的排序是一种罕见的问题，现在，因为我们都知道在创建商品或答案之前应该调用 addStream ;即使是重新谈判的会议。

Ordering of the code is, kind of rare-issues, nowadays, because we all know that addStream should be called before creating offer or answer; even for renegotiated sessions.

尝试使用 chrome：// webrtc-internals 和Firefox的 about：config 查看这些浏览器中发生的事情;并始终使用控制台日志进行 onIceConnectionStateChange 事件，以帮助您在ICE连接检查中检查ICE-Agent 是否失败进程或...

Try to use chrome://webrtc-internals and Firefox's about:config to see what's happening inside these browsers; and always use console-logs for onIceConnectionStateChange event which helps you check if ICE-Agent failed in the ICE-connectivity check process or...

有时候，对于提议者来说，设置-remote-sdp会导致异常。始终使用 onSdpError 同时为 createOffer / createAnswer 和 setLocalDescription / setRemoteDescription 例如

Sometimes setting-remote-sdp for offerer too earlier, causes exceptions. Always use onSdpError both for createOffer/createAnswer and setLocalDescription/setRemoteDescription e.g.

peer .setRemoteDescription（remoteSDP，onSdpSuccess，onSdpFailure）;


一些演示资源：

1. /

1. https://github.com/muaz-khan/WebRTC-Experiment / Demos
2. https://github.com/mozilla/webrtc-landing

和

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