Re: [Live-devel] RTP over TCP video stream break problem

> On Aug 9, 2023, at 8:03 PM, Андрій Олексійович Радченко > wrote: > > The problem isn't with the network, but with the likelihood that the client > or even the network itself might glitch. Can we really expect a system that's > working fine now to continue operating smoothly in the coming d

Re: [Live-devel] RTP over TCP video stream break problem

The problem isn't with the network, but with the likelihood that the client or even the network itself might glitch. Can we really expect a system that's working fine now to continue operating smoothly in the coming days? And if such a glitch occurs, we get a closed stream instead of a recovery, ev

Re: [Live-devel] RTP over TCP video stream break problem

> On Aug 9, 2023, at 6:14 PM, Андрій Олексійович Радченко > wrote: > > Yes, but there is no answer why live555 has to close the stream when it can't > send it over TCP. The comments in the code for “sendDataOverTCP()” explain this very clearly: // The blocking "send()" failed, or ti

Re: [Live-devel] RTP over TCP video stream break problem

Yes, but there is no answer why live555 has to close the stream when it can't send it over TCP. Other servers, as I understand, don't do it. And clients can recover from inconsistent data. So, maybe I misunderstood something, but I see no reason for such behaviour. Can you please explain it to me?

Re: [Live-devel] RTP over TCP video stream break problem

> On Aug 9, 2023, at 5:37 PM, Андрій Олексійович Радченко > wrote: > > Hi. I use live555 to create server and client that stream and get video and > audio respectively. I use RTP over TCP due to possible bad network. If use > UDP instead of TCP then in some situations can't get I-frames of H

[Live-devel] RTP over TCP video stream break problem

Hi. I use live555 to create server and client that stream and get video and audio respectively. I use RTP over TCP due to possible bad network. If use UDP instead of TCP then in some situations can't get I-frames of H264 stream and nothing is decoding, showing etc, I can't afford this in my system.

Re: [Live-devel] RTP-over-TCP error

> On Nov 26, 2020, at 11:30 PM, Лисовой Андрей Витальевич > wrote: > > Hi! > > I found bug (using live.2018.02.18, but in 2020.06.25 RTPInterface.cpp is the > same). First, you should always try to use the latest version of the code (currently, 2020.11.26) > It occures when proxyServer u

[Live-devel] RTP-over-TCP error

Hi! I found bug (using live.2018.02.18, but in 2020.06.25 RTPInterface.cpp is the same). It occures when proxyServer used with -t flag (RTP-over-TCP with backend server). In some cases, like client disconnects or reconnects, backend socket deregistering RTP channels, and only RTCP channels have

Re: [Live-devel] RTP over TCP streaming

2016-03-23 16:15 GMT+01:00 Ross Finlayson : > > I have some doubts about how to stream RTP over TCP. > > > > I am trying to do so without RTSP > > Why? The whole point of the RTP-over-TCP implementation is that it uses > the existing TCP connection that’s used to send RTSP commands. > Ok I see,

Re: [Live-devel] RTP over TCP streaming

> I have some doubts about how to stream RTP over TCP. > > I am trying to do so without RTSP Why? The whole point of the RTP-over-TCP implementation is that it uses the existing TCP connection that’s used to send RTSP commands. It might conceivably be possible to make RTP-over-TCP work with s

[Live-devel] RTP over TCP streaming

Dear all, I have some doubts about how to stream RTP over TCP. I am trying to do so without RTSP, setting manually the IP and Ports, as I do for UDP in a code like the following: rtpGroupsock = new Groupsock(*env, destinationAddress, Port(port), TTL); sink = H264VideoRTPSink::createNew(*env, rtp

Re: [Live-devel] rtp over tcp

> 1)When I use centos's default value of net.core.rmem_max > /net.core.wmem_max/net.core.wmem_default,the "LIVE555 Streaming Media" shows > some information below and then openRTSP can't receive any data. > sendDataOverTCP: resending 970 byte send (blocking) > sendDataOverTCP: resending 300 byt

Re: [Live-devel] rtp over tcp

-What OS are you running (for both your client (VLC) and server)? A:The server OS is centos6.5_x64,the client OS is WIN7 - Are you using the latest version of the "LIVE555 Streaming Media" code (the only version that we support) for your server? A:yes.The version of the "LIVE555 Streaming Media"

Re: [Live-devel] rtp over tcp

> When I transport rtp packets over tcp using live555 , I get many logs like > below after playing many minutes. > sendRTPorRTCPPacketOverTCP: failed! (errno 32) > sendRTPorRTCPPacketOverTCP: failed! (errno 32) > ... > > or > > sendRTPorRTCPPacketOverTCP: failed! (errno 9) > sendRTPorRTCPPacke

[Live-devel] rtp over tcp

When I transport rtp packets over tcp using live555 , I get many logs like below after playing many minutes. sendRTPorRTCPPacketOverTCP: failed! (errno 32) sendRTPorRTCPPacketOverTCP: failed! (errno 32) ... or sendRTPorRTCPPacketOverTCP: failed! (errno 9) sendRTPorRTCPPacketOverTCP: failed! (e

Re: [Live-devel] RTP over TCP packet size determination

> If I changed the value to smaller one, like (256, 400), the packet size keeps > the same size(~1300). Calling "setPacketSizes(256,400)" on the newly-created "MultiFramedRTPSink" object should work. You shouldn't modify the supplied code. Ross Finlayson Live Networks, Inc. http://www.live55

[Live-devel] RTP over TCP packet size determination

Hi, I sent the audio and video muxed in MPEG2TS container to an mobile phone(Android) via live555. The traffic from Live555 server to phone is via RTP over TCP, for the mobile phone is behind a NAT gateway. It goes well until I was request to send audio data only. Because that the bitrate for a

Re: [Live-devel] RTP over TCP

} break; } From: live-devel-boun...@ns.live555.com [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson Sent: Monday, October 01, 2012 12:24 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] RTP over TCP To prevent this situation,

Re: [Live-devel] RTP over TCP

> To prevent this situation, I tried adding a check to > SocketDescriptor::tcpReadHandler1() in the AWAITING_DOLLAR case to prevent it > from calling fServerRequestAlternativeByteHandler if the byte read was 0xFF > or 0xFE. This is, of course, a hack on top of a hack. Yes, but since the origin

[Live-devel] RTP over TCP

I'm streaming RTP data over TCP from a network camera. Most of the time this works fine. But about once every 30 minutes (varies a lot) this particular camera sends some unexpected data between two RTP packets and that causes live555 to go into a state where it thinks it's receiving RTSP data

Re: [Live-devel] RTP over TCP - What happens if a client closes the RTP socket gracelessly?

> If an RTSP client (not developed by me and sadly not based on Live555) > asks for a video, streaming with RTP over TCP, and after some minutes closes > the RTP socket gracelessly (bug or black-out), doesn't (or can't) send the > RTSP Teardown command, then the Live555 RTSP Server > (OnDem

[Live-devel] RTP over TCP - What happens if a client closes the RTP socket gracelessly?

Hello Ross. If an RTSP client (not developed by me and sadly not based on Live555) asks for a video, streaming with RTP over TCP, and after some minutes closes the RTP socket gracelessly (bug or black-out), doesn't (or can't) send the RTSP Teardown command, then the Live555 RTSP Server (On

[Live-devel] RTP over TCP bugs

Hi Ross, Thanks for your response. >I'm not interested in any alleged server bug, unless the server is >running an up-to-date version of our software (in which case it >doesn't matter what version of our software the client is running, or >whether or not the client is using the synchronous or asy

Re: [Live-devel] RTP over TCP bugs

Could you please clarify this a bit? If I understand correctly, we must therefore upgrade both our server and client code to use the latest live media version to avoid the server hanging? Or would updating the client code to use the new async RTSPClient code suffice? I'm not interested in any al

[Live-devel] RTP over TCP bugs

In case it helps: profiling shows that the following seems to account for the CPU usage: void SocketDescriptor::tcpReadHandler(SocketDescriptor* socketDescriptor, int mask) { socketDescriptor->tcpReadHandler1(mask); } Commenting out the dummy descriptor on fReadSet didn't help solve the issue.

[Live-devel] RTP over TCP bugs

Hi Ross, We use the LIVE555 Streaming Media library to implement a live streaming server (on Ubuntu) and a windows media player RTSP client streaming RTP over TCP only. We have run into a complicated set of problems recently: 1) When a client with a *bad* internet connection joins the RTSP server

Re: [Live-devel] RTP over TCP with iPhone

Dmitrij Gesendet: Donnerstag, 30. Juni 2011 09:40 An: LIVE555 Streaming Media - development & use Betreff: Re: [Live-devel] RTP over TCP with iPhone I can't change the code incremental. I have to use a new/ohther sink. And it works fine if I invoke this code without &q

Re: [Live-devel] RTP over TCP with iPhone

layson Gesendet: Donnerstag, 30. Juni 2011 09:18 An: LIVE555 Streaming Media - development & use Betreff: Re: [Live-devel] RTP over TCP with iPhone >if I implement it in this way i never get into "afterGettingFrame". >What am I doing wrong? I don't know; you're going to ha

Re: [Live-devel] RTP over TCP with iPhone

if I implement it in this way i never get into "afterGettingFrame". What am I doing wrong? I don't know; you're going to have to figure this out for yourself. Remember, you're starting from code that you know works - the "openRTSP" code. By making incremental changes to this code, and testin

Re: [Live-devel] RTP over TCP with iPhone

Yes I use FileSink as model: class MyMediaSink:public MediaSink { public: MyMediaSink(UsageEnvironment &env):MediaSink(env){buffer = new uint8_t[fileSinkBufferSize];} virtual ~MyMediaSink(){} protected: Boolean continuePlaying() { if (NULL == fSource)

Re: [Live-devel] RTP over TCP with iPhone

Boolean continuePlaying(){/*CODE will come soon*/ return true;} I'm not sure what your question is, but once your "MyMediaSink" class should work once you implement the "continuePlaying()" function (etc.). I suggest that you use the "FileSink" code as a model. -- Ross Finlayson Live

Re: [Live-devel] RTP over TCP with iPhone

ch, 29. Juni 2011 16:24 An: LIVE555 Streaming Media - development & use Betreff: Re: [Live-devel] RTP over TCP with iPhone Hi Ross, I followed your advice and start to implement with openRTSP. I need data in buffer and not in file and so i wrote little class to subclass MediaSink:

Re: [Live-devel] RTP over TCP with iPhone

Hi Ross, I followed your advice and start to implement with openRTSP. I need data in buffer and not in file and so i wrote little class to subclass MediaSink: class MyMediaSink:public MediaSink { public: MyMediaSink(UsageEnvironment &env):MediaSink(env){buffer = new uint8_t();}

Re: [Live-devel] RTP over TCP with iPhone

> You already know that "openRTSP -t" - built from the supplied, unmodified source code - works OK on your system. I have to correct my statement. Sorry! I get only first image with: ./openRTSP -t -F __Video rtsp://193.158.57.66:8555/mjpeg You probably got a very large video file, wh

Re: [Live-devel] RTP over TCP with iPhone

Am 16.06.2011 um 11:10 schrieb Ross Finlayson: > Please don't send the same message to the mailing list more than once. > Sorry for this! > You already know that "openRTSP -t" - built from the supplied, unmodified > source code - works OK on your system. I have to correct my statement. Sor

Re: [Live-devel] RTP over TCP with iPhone

Please don't send the same message to the mailing list more than once. You already know that "openRTSP -t" - built from the supplied, unmodified source code - works OK on your system. So I suggest that you start from this code, making only small, incremental changes at a time, until it evolve

Re: [Live-devel] RTP over TCP with iPhone

Hello Ross, I am still searching for my error. I had debugged in your source code in the hope that i can find my error. I came to the place where you read data from TCP. In RTPInterface.cpp in 'tcpReadHandler1' I figure out, that the value of 'fNextTCPReadStreamSocketNum' is a big negative

Re: [Live-devel] RTP over TCP with iPhone

Hello Ross, I am still searching for my error. I had debugged in your source code in the hope that i can find my error. I came to the place where you read data from TCP. In RTPInterface.cpp in 'tcpReadHandler1' I figure out, that the value of 'fNextTCPReadStreamSocketNum' is a big negative

Re: [Live-devel] RTP over TCP with iPhone

I guess that it isn't a server's problem. I tested the connection with an unmodified openRTSP client and got an image(see output below). My question is, are there other places i have to change in source code for using TCP except sendSetupCommand? No. -- Ross Finlayson Live Networks, Inc. http

Re: [Live-devel] RTP over TCP with iPhone

Hello Ross, thank you for replying! I guess that it isn't a server's problem. I tested the connection with an unmodified openRTSP client and got an image(see output below). My question is, are there other places i have to change in source code for using TCP except sendSetupCommand? Thanks!

Re: [Live-devel] RTP over TCP with iPhone

i am developing an iPhone/iPad App that receives RTSP-streams. It works fine, while the server and client are in the same network. But If I try to get a stream from outside(from Internet) i run in problems with UDP-ports. On the server side i can open ports, but on a client side it is sometimes

Re: [Live-devel] RTP over TCP problem in the latest version of library

I have a few problems. I downloaded the latest version of LiveMedia library 2010.06.22 and RTPoverTCP streaming ( RTSPClient ) suddenly stopped working. In the previous version I tested - 2010.06.11 everything was working ok. Please send us the (publically-accessible) "rtsp://" URL of a server

[Live-devel] RTP over TCP problem in the latest version of library

Hello, I have a few problems. I downloaded the latest version of LiveMedia library 2010.06.22 and RTPoverTCP streaming ( RTSPClient ) suddenly stopped working. In the previous version I tested - 2010.06.11 everything was working ok. PLAY command is invoked correctly by RTSPClient but no frames

[Live-devel] RTP over TCP, pushing

Is it possible for Live555 to push a unicast stream to a server online, and have all the traffic go over TCP? I know Live555 can interleave AV data over the RTCP TCP channel, but I've only seen that for the on-demand configurations; is it possible to "push" video to a server over the RTCP line? T

Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel)

Ross Finlayson: > In the meantime, if you want to use RTP-over-TCP streaming with our > server, then you should *not* send RTSP commands as 'keep alive' > indicators. Instead, you *must* send RTCP "RR" packets. Note that > RTCP is a *mandatory* part of the RTP specification, so if your > clie

Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel)

As I have noted in the past (several times now), the issue with the server handling incoming RTSP commands (after the initial "PLAY"), if RTP-over-TCP streaming is used, is a known problem. When this is solved, I will let this mailing list know. In the meantime, if you want to use RTP-over-TCP

Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel)

Ross Finlayson: > This is not relevant to our implementation, because our server also > uses RTCP "RR" packets - coming from the client - as a keep-alive, > and clients (including ours) send those far more frequently than once > every 80 seconds. > > If you have a client that is properly sendin

Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel)

Yes its true, the RTCP "RR" packets are far more frequent than the session timeout, they appear about once every five seconds. For UDP streams that worked OK but for some reason on TCP streams, adding the session timeout did the trick. In the case of RealPlayer, UDP worked fine but on TCP it alm

Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel)

>I'd appreciate any input on the meaning of the session timeout >field, its recommended value (80?). This is not relevant to our implementation, because our server also uses RTCP "RR" packets - coming from the client - as a keep-alive, and clients (including ours) send those far more frequently

Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel)

Thanks for the replies sent on the subject of RTP-over-TCP I've looked into this issue and have managed to get good results. The original problem I had was using testOnDemandRTSPServer example to play a stream of of mpeg4 encodered frames (elementary stream) via TCP (RTP over TCP). The video

Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel)

To best of my knowledge, Live555 does not yet support that RTSP tunneled over HTTP Yes it does - in our *client* implementation. Note, for example, the "-T " option to "openRTSP": Our RTSP *server* implementation, however, does not yet suppor

Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel)

AIL PROTECTED] On Behalf Of Ross Finlayson Sent: Wednesday, October 03, 2007 8:40 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel) >Received RTCP "BYE" on "video/MP4V-ES" subsession (after 672 seconds) T

Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel)

>Received RTCP "BYE" on "video/MP4V-ES" subsession (after 672 seconds) The RTCP "BYE" packet (from the server) is your server is explicitly saying "I'm ending the streaming of the stream now". The client then (correctly) handles this by shutting down. So, you will need to figure out why your s

Re: [Live-devel] RTP-over-TCP streaming (Noam Camiel)

system interface to select(). Ron McOuat Message: 3 Date: Tue, 2 Oct 2007 17:10:25 -0700 (PDT) From: Noam Camiel <[EMAIL PROTECTED]> Subject: [Live-devel] RTP-over-TCP streaming To: LIVE555 Streaming Media - development & use <[EMAIL PROTECTED]> Message-ID: <[EMA

Re: [Live-devel] RTP-over-TCP streaming

>Should I look for specific packets in RTP-over-TCP that are some >sort of keep-alive? Yes, the client (should) send periodic RTCP "RR" packets - for both RTP/UDP and RTP/TCP. VLC and QuickTime Player should send these; for RealPlayer, I can't say for sure. -- Ross Finlayson Live Networks, I

Re: [Live-devel] RTP-over-TCP streaming

> I don't know of any issue with the code that could be causing your > problem. When you stream RTP-over-TCP, are you doing so using the > same network as RTP-over-UDP? If not, then perhaps there's something > (e.g., a firewall) in the middle of your RTP-over-TCP network that's > causing you

Re: [Live-devel] RTP-over-TCP streaming

>I have a problem when accessing live server's stream using >RTP-over-TCP as opposed to UDP. > >I have used the testOnDemandRTSPServer example to play a stream of >of mpeg4 encodered frames (elementary stream) and was successful >with both VLC and Quicktime as well as RealPlayer, using unicast

[Live-devel] RTP-over-TCP streaming

Hello I have a problem when accessing live server's stream using RTP-over-TCP as opposed to UDP. I have used the testOnDemandRTSPServer example to play a stream of of mpeg4 encodered frames (elementary stream) and was successful with both VLC and Quicktime as well as RealPlayer, using unicast

Re: [Live-devel] RTP over TCP sometimes locks up the server

Peter, Thanks for the report. Yes, you're correct that the current code is buggy - it should not be doing more than one potentially blocking socket read inside the read handler (because the second and subsequent reads might, indeed, block). It seems a bit strange that a client would behave i

[Live-devel] RTP over TCP sometimes locks up the server

In live555 release 0.18 (1st July 2007), I have occassional found that the server can "appear" to lock up when streaming RTP over TCP. After much analysis I think I have traced the problem to the tcpReaderHandler method in RTP_Interface.cpp. Specifically lines 332 to 337 do { if (readSoc