How could I get the bytes send amount and receive amount that shows as chrome?
I want to use the data to check the connection quality. Or are there any suggestion that I should consider?
Use the getStats API. https://webrtc.github.io/samples/src/content/peerconnection/bandwidth/ shows an example.
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I am using pyshark to analyze a Wireshark capture. As you can see in the screenshot there is a packet containing two MMS subpackets. But when I load capture with wireshark and watch the pyshark representation of thepacket with a debugger, there is only the data of the first MMS packet.
Can anyone tell me how I can access the data of the second subpacket?
I have been trying to figure out a way to calculate the following:
Bandwidth, Latency, Current Upload, and Download speed.
And have been confused with the values I am getting for the INBOUND-RTP, OUTBOUND-RTP, & REMOTE-INBOUND-RTP.
In my head, I was thinking about inbound-rtp as a collection of stats for all incoming data.
which apparently is wrong, since different stats for that type always stays Zero
The current setup uses chrome as a 2 connecting Clients, and a Media Server, with client instances running on "localhost"
The terminology used on MDN is a bit terse, so here's a rephrasing that I hope is helpful to solve your problem! Block quotes taken from MDN & clarified below. For an even terser description, also see the W3C definitions.
outbound-rtp
An RTCOutboundRtpStreamStats object giving statistics about an outbound RTP stream being sent from the RTCPeerConnection.
This stats report is based on your outgoing data stream to your peers. This is the measurement taken from the perspective of just that oubound RTP stream, which is why information that involves your peers (round trip time, jitter, etc.) is missing, because those can only be measured with an understanding of the peer's processing of your stream.
inbound-rtp
Statistics about an inbound RTP stream that's currently in use by this RTCPeerConnection, in an RTCInboundRtpStreamStats object.
By contrast to the Outbound RTP statistics, this stats report contains data about the inbound data stream you are receiving from your peer(s). Notice that if you do not have any connected peers your call to getStats does not include this report type at all.
remote-inbound-rtp
Contains statistics about the remote endpoint's inbound RTP stream; that stream corresponds to the local endpoint's outbound RTP stream. Using this RTCRemoteInboundRtpStreamStats object, you can learn how the well the remote peer is receiving data.
This stats report provides details about your outbound rtp stream from the perspective of the remote connection. That is to say that this stats report provides an analysis about your outbound-rtp stream from the perspective of the remote server that is handling your stream on the other side.
I'm on the MDN writing team at Mozilla and happened upon this just now. I've taken some of the information from this conversation and applied it back to the article about RTCStatsType. There's more to improve there still, but I wanted to thank you for that insight!
Always feel free to sign up for an MDN account and update any content you see that's inaccurate or incomplete! Or you can file an issue and we'll see what we can do.
I am currently working on an application to change my RGBWW light strips by a Java application.
Information has to be sent via UDP packages in order to be understood by the controller.
Unfortunately, the hex number 0x80 has to be sent - which is causing some problems.
Whenever I send a byte array containing only numbers fron 0x00 to 0x79 (using DataPacket and a DataSocket), I do get an UDP Package popping up on my network monitor.
As soon as I include the number 0x80 or any other higher, I see two things Happen:
1: I do not longer get only UDP protocols, but messages are displayed as RTP / RTCP most of the time
2: The method Integer.hexToString() does not display "80", but gives me a "ffffff80".
My question: Is there something I am missing when it comes to sending hex info by UDP? Or is there another way of sending it, possibly avoiding the annoyingly signed bytes?
I unfortunately did not find any information that would have significantly helped me on that issue, but I hope you can help me!
Thanks in advance!
I am trying to build my own client RTMP library for an app that I am working on. So far everything has gone pretty successfully in that I am able to connect to the RTMP server negotiate the handshake and then send all the necessary packets (FCPublish Publish ETC) then from the server i get the onStatus message of NetStream.Publish.Start which means that I have successfully got the server to allow me to start publishing my live video broadcast. Wireshark also confirms that the information (/Data packetizing) is correct as it shows up correctly there also.
Now for where I am having some trouble is RTMP Chunking, going off the Adobe RTMP Specification on page 17 & 18 shows an example of how a message is chunked. From this example I can see that it is broken down based on the chunk size (128 bytes). For me the chunk size gets negotiated in the initial connect and exchange which is always 4096 bytes. So for when I am exchanging video data that is larger than 4096 bytes I need to chunk the message down sending the RTMP packetHeader combined with the first 4096 bytes of data then sending a small RTMP header which is 0xc4 (0xc0 | packetHeaderType (0x04)) combined with 4096 bytes of video data until the full packet specified by the header has been sent. Then a new frame comes in and the same process is repeated.
By checking other RTMP client example written in different languages this seems to be what they are all doing. Unfortunately the ingest server that I am trying to stream to is not picking up the broadcast video data, they dont close the connection on my they just never show video or any sign that the video is right. Wireshark shows that after the video atom packet is sent most packets sent are Unknown (0x0) for a little bit and then they will switch into Video Data and will sort of flip flop between showing Unknown (0x0) and Video Data. However if I restrict my payload max size to 20000 bytes Wireshark shows everything as Video Data. Obviously the ingest server will not show video in this situation as i am removing chunks of data for it to be only 20k bytes.
Trying to figure out what is going wrong I started another xcode project that allows me to spoof a RTMP server on my Lan so that I can see what the data looks like from libRTMP IOS as it comes into the server. Also with libRTMP I can make it log the packets it sends and they seem to inject the byte 0xc4 even 128 bytes even tho I have sent the Change Chunk size message as the server. When I try to replicate this in my RTMP client Library by just using a 128 chunk size even tho it has been set to 4096 bytes the server will close my connection on me. However if change libRTMP to try to go to the live RTMP server it still prints out within LibRTMP that it is sending packets in a chunk size of 128. And the server seems to be accepting it as video is showing up. When I do look at the data coming in on my RTMP server I can see that it is all their.
Anyone have any idea what could be going on?
While I haven't worked specifically with RTMP, I have worked with RTSP/RTP/RTCP pretty extensively, so, based on that experience and the bruises I picked up along the way, here are some random, possibly-applicable tips that might help/things to look for that might be causing an issue:
Does your video encoding match what you're telling the server? In other words, if your video is encoded as H.264, is that what you're specifying to the server?
Does the data match the container format that the server is expecting? For example, if the server expects to receive an MPEG-4 movie (.m4v) file but you're sending only an encoded MPEG-4 (.mp4) stream, you'll need to encapsulate the MPEG-4 video stream into an MPEG-4 movie container. Conversely, if the server is expecting only a single MPEG-4 video stream but you're sending an encapsulated MPEG-4 Movie, you'll need to de-mux the MPEG-4 stream out of its container and send only that content.
Have you taken into account the MTU of your transmission medium? Regardless of chunk size, getting an MTU mismatch between the client and server can be hard to debug (and is possibly why you're getting some packets listed as "Unknown" type and others as "Video Data" type). Much of this will be taken care of with most OS' built-in Segmentation-and-Reassembly (SAR) infrastructure so long as the MTU is consistent, but in cases where you have to do your own SAR logic it's very easy to get this wrong.
Have you tried capturing traffic in Wireshark with libRTMP iOS and your own client and comparing the packets side by side? Sometimes a "reference" packet trace can be invaluable in finding that one little bit (or many) that didn't originally seem important.
Good luck!
I am building an Arduino-based device that needs to send data over the internet to a remote server. It needs to do this as frequently as possible but also use as little bandwidth as possible. It will probably work over GSM/EDGE (cellular networking).
The data I'd like to send is about 40 bytes in size - really minimal. I'd like to send this packet to the server about once a minute, but also receive a packet of around that size in response once in a while.
The bandwidth on my server is no problem - the bandwidth on the device's internet connection is, i.e. the cellular data.
Do headers on mobile requests and responses count as part of the bandwidth?
Yes, the total packet size is all data that is sent. Assuming a TCP packet you lose 20 bytes right from the start. If you get intimate with Wireshark you can see exactly what's happening.