February 2011 - rtmpdump - lists.mplayerhq.hu

New handshake?
by Blue Cop 12 Mar '11

12 Mar '11

I noticed some limelight rtmp servers using what seems to be a new handshake type. here is teh -z output. http://pastie.org/1606602 some limelight servers still work. the ones that do answer 8 WARNING: HandShake: Type mismatch: client sent 6, server answered 8 while all the ones that don't answer 9 WARNING: HandShake: Type mismatch: client sent 6, server answered 9 I am not sure if this is news to you guys but just wanted to let you know about it.

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rtmpdump, FMS and NetStream::send issue
by Alexander Ivanenko 28 Feb '11

28 Feb '11

Hi All, I have a compatibility issue with rtmpdump 2.3 and fms 3.5/4.0. The trouble is beacause of the 3 lines in ActionScript publisher my client uses: -------------------- ... // publish the stream by name nsPublish.publish(publishName.text); // Here they are: // add custom metadata to the stream var metaData:Object = new Object(); metaData["description"] = "Chat using VideoChat example." nsPublish.send("@setDataFrame", "onMetaData", metaData); // attach the camera and microphone to the server nsPublish.attachCamera(camera); nsPublish.attachAudio(microphone); ... ------------ When I try to read the stream published with metadata using rtmpdump --verbose --live -r rtmp://127.0.0.1/live/myStream -o 1.flv rtmpdump fails with log listed below. However, any native flash-player plays the stream ok. I would like to make a patch that fixes an issue, but don't even know where to start. It seems like FMS just closes the pipe at some moment. Please help, any suggestions are appreciated. Alexander. rtmpdump log: [alex@romul rtmpdump-2.3]$ rtmpdump --verbose --live -r rtmp://127.0.0.1/live/myStream -o 1.flv RTMPDump v2.3 (c) 2010 Andrej Stepanchuk, Howard Chu, The Flvstreamer Team; license: GPL DEBUG: Parsing... DEBUG: Parsed protocol: 0 DEBUG: Parsed host : 127.0.0.1 DEBUG: Parsed app : live DEBUG: Protocol : RTMP DEBUG: Hostname : 127.0.0.1 DEBUG: Port : 1935 DEBUG: Playpath : myStream DEBUG: tcUrl : rtmp://127.0.0.1:1935/live DEBUG: app : live DEBUG: live : yes DEBUG: timeout : 30 sec DEBUG: Setting buffer time to: 36000000ms Connecting ... DEBUG: RTMP_Connect1, ... connected, handshaking DEBUG: HandShake: Type Answer : 03 DEBUG: HandShake: Server Uptime : 568712 DEBUG: HandShake: FMS Version : 3.5.1.1 DEBUG: HandShake: Handshaking finished.... DEBUG: RTMP_Connect1, handshaked DEBUG: Invoking connect INFO: Connected... DEBUG: HandleServerBW: server BW = 2500000 DEBUG: HandleClientBW: client BW = 2500000 2 DEBUG: HandleCtrl, received ctrl. type: 0, len: 6 DEBUG: HandleCtrl, Stream Begin 0 DEBUG: RTMP_ClientPacket, received: invoke 240 bytes DEBUG: 0000: 02 00 07 5f 72 65 73 75 6c 74 00 3f f0 00 00 00 ..._result.?.... DEBUG: 0010: 00 00 00 03 00 06 66 6d 73 56 65 72 02 00 0d 46 ......fmsVer...F DEBUG: 0020: 4d 53 2f 33 2c 35 2c 31 2c 35 31 36 00 0c 63 61 MS/3,5,1,516..ca DEBUG: 0030: 70 61 62 69 6c 69 74 69 65 73 00 40 3f 00 00 00 pabilities.@?... DEBUG: 0040: 00 00 00 00 04 6d 6f 64 65 00 3f f0 00 00 00 00 .....mode.?..... DEBUG: 0050: 00 00 00 00 09 03 00 05 6c 65 76 65 6c 02 00 06 ........level... DEBUG: 0060: 73 74 61 74 75 73 00 04 63 6f 64 65 02 00 1d 4e status..code...N DEBUG: 0070: 65 74 43 6f 6e 6e 65 63 74 69 6f 6e 2e 43 6f 6e etConnection.Con DEBUG: 0080: 6e 65 63 74 2e 53 75 63 63 65 73 73 00 0b 64 65 nect.Success..de DEBUG: 0090: 73 63 72 69 70 74 69 6f 6e 02 00 15 43 6f 6e 6e scription...Conn DEBUG: 00a0: 65 63 74 69 6f 6e 20 73 75 63 63 65 65 64 65 64 ection succeeded DEBUG: 00b0: 2e 00 0e 6f 62 6a 65 63 74 45 6e 63 6f 64 69 6e ...objectEncodin DEBUG: 00c0: 67 00 00 00 00 00 00 00 00 00 00 04 64 61 74 61 g...........data DEBUG: 00d0: 08 00 00 00 00 00 07 76 65 72 73 69 6f 6e 02 00 .......version.. DEBUG: 00e0: 09 33 2c 35 2c 31 2c 35 31 36 00 00 09 00 00 09 .3,5,1,516...... DEBUG: (object begin) DEBUG: (object begin) DEBUG: Property: <Name: fmsVer, STRING: FMS/3,5,1,516> DEBUG: Property: <Name: capabilities, NUMBER: 31.00> DEBUG: Property: <Name: mode, NUMBER: 1.00> DEBUG: (object end) DEBUG: (object begin) DEBUG: Property: <Name: level, STRING: status> DEBUG: Property: <Name: code, STRING: NetConnection.Connect.Success> DEBUG: Property: <Name: description, STRING: Connection succeeded.> DEBUG: Property: <Name: objectEncoding, NUMBER: 0.00> DEBUG: Property: <Name: data, OBJECT> DEBUG: (object begin) DEBUG: Property: <Name: version, STRING: 3,5,1,516> DEBUG: (object end) DEBUG: (object end) DEBUG: (object end) DEBUG: HandleInvoke, server invoking <_result> DEBUG: HandleInvoke, received result for method call <connect> DEBUG: sending ctrl. type: 0x0003 DEBUG: Invoking createStream DEBUG: FCSubscribe: myStream DEBUG: Invoking FCSubscribe DEBUG: RTMP_ClientPacket, received: invoke 29 bytes DEBUG: 0000: 02 00 07 5f 72 65 73 75 6c 74 00 40 00 00 00 00 ..._result.@.... DEBUG: 0010: 00 00 00 05 00 3f f0 00 00 00 00 00 00 .....?....... DEBUG: (object begin) DEBUG: Property: NULL DEBUG: (object end) DEBUG: HandleInvoke, server invoking <_result> DEBUG: HandleInvoke, received result for method call <createStream> DEBUG: SendPlay, seekTime=0, stopTime=0, sending play: myStream DEBUG: Invoking play DEBUG: sending ctrl. type: 0x0003 Caught signal: 13, cleaning up, just a second... ERROR: WriteN, RTMP send error 32 (42 bytes) ERROR: RTMP_ReadPacket, failed to read RTMP packet header DEBUG: Closing connection.

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rtmpte xbmc python
by Sen Heng 26 Feb '11

26 Feb '11

Hi Guys, I try to use XBMC play RTE player. I can success dump the video, but not sure how to use python to write swf verification. I would very much appreciate if some one can give me an example for below link use python. rtmpdump -r "rtmpte://fmsod.rte.ie/rtevod/" -a "rtevod/" -a "rtevod/" -f "WIN 10,1,82,76" -W " http://www.rte.ie/player/assets/player_417.swf?shareBtn=0" -p " http://www.rte.ie/player/#v=1092590" -y "mp4:/2011/0224/20110224rte1a_20110224_192806_part1_eastenders__10__2010_001255.mp4" -o anything.mp4 Kind regards, Sen

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[PATCH] Publisher authentication
by Steve McFarlin 25 Feb '11

25 Feb '11

The attached adobe auth patch has an infinite recursion in it. I believe it is the one from Sergiy. Stack Trace: .. ... #743 0x0047a818 in RTMP_ConnectStream at rtmp.c:1047 #744 0x0047c2c6 in HandleInvoke at rtmp.c:2765 #745 0x0047ae92 in RTMP_ClientPacket at rtmp.c:1248 #746 0x0047a818 in RTMP_ConnectStream at rtmp.c:1047 #747 0x0047c2c6 in HandleInvoke at rtmp.c:2765 #748 0x0047ae92 in RTMP_ClientPacket at rtmp.c:1248 #749 0x0047a818 in RTMP_ConnectStream at rtmp.c:1047 #750 0x0047c2c6 in HandleInvoke at rtmp.c:2765 #751 0x0047ae92 in RTMP_ClientPacket at rtmp.c:1248 #752 0x0047a818 in RTMP_ConnectStream at rtmp.c:1047 The above file offsets may be different then a auto patched librtmp. I had to manually patch. This occurs when connecting to a Wowza Media Server using an invalid application for publishing. The server rejects the connection and then the following block in HandleInvoke initiates the infinite recursion. Specifically the call to RTMP_ConnectStream. I have not had a chance to fix this. If I patch the patch I will post it. else if (AVMATCH(&method, &av_close)) { RTMP_Log(RTMP_LOGERROR, "rtmp server requested close"); RTMP_Close(r); #ifdef CRYPTO if ((r->Link.protocol & RTMP_FEATURE_WRITE) && !(r->Link.pFlags & RTMP_PUB_CLEAN) && ( !(r->Link.pFlags & RTMP_PUB_NAME) || !(r->Link.pFlags & RTMP_PUB_RESP) || (r->Link.pFlags & RTMP_PUB_CLATE) ) ) { /* clean later */ if(r->Link.pFlags & RTMP_PUB_CLATE) r->Link.pFlags |= RTMP_PUB_CLEAN; RTMP_Log(RTMP_LOGERROR, "authenticating publisher"); if (!RTMP_Connect(r, NULL) || !RTMP_ConnectStream(r, 0)) goto leave; } #endif } steve

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getting live streams from tvpc.com
by Phoenix Revived 25 Feb '11

25 Feb '11

I have been trying unsuccessfully to get the live streams from www.tvpc.com (e.g http://www.tvpc.com/Channel.php?ChannelID=2686) using rtmpsrv and rtmpsuck. Anybody have sucess with this? Thanks!

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r557 - trunk/librtmp/rtmp.c
by hyc 25 Feb '11

25 Feb '11

Author: hyc Date: Fri Feb 25 06:51:40 2011 New Revision: 557 Log: Fix FLV timestamps during resume Modified: trunk/librtmp/rtmp.c Modified: trunk/librtmp/rtmp.c ============================================================================== --- trunk/librtmp/rtmp.c Sun Feb 20 03:15:32 2011 (r556) +++ trunk/librtmp/rtmp.c Fri Feb 25 06:51:40 2011 (r557) @@ -4083,7 +4083,7 @@ Read_1_Packet(RTMP *r, char *buf, unsign /* grab first timestamp and see if it needs fixing */ nTimeStamp = AMF_DecodeInt24(packetBody + 4); nTimeStamp |= (packetBody[7] << 24); - delta = packet.m_nTimeStamp - nTimeStamp; + delta = packet.m_nTimeStamp - nTimeStamp + r->m_read.nResumeTS; while (pos + 11 < nPacketLen) {

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[PATCH] - Compiling under MinGW with latest PolarSSL
by faltu faltu 21 Feb '11

21 Feb '11

Hi Developers, I recently compiled rtmpdump.exe under windows with MinGW. However to compile librtmp.dll "-mdll" switch was required to pass to the linker otherwise it complained about "Undefined reference to WinMain". Also name of the function "set_default_ciphers" is changed to "set_default_ciphersuites" alongwith it'ss parameter name in latest PolarSSL release. Patch.diff file is attached herewith for your consideration. Thanks.

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r556 - web/index.html
by hyc 20 Feb '11

20 Feb '11

Author: hyc Date: Sun Feb 20 03:15:32 2011 New Revision: 556 Log: Add TV3 project Modified: web/index.html Modified: web/index.html ============================================================================== --- web/index.html Tue Dec 28 15:59:45 2010 (r555) +++ web/index.html Sun Feb 20 03:15:32 2011 (r556) @@ -138,6 +138,7 @@ Projects using rtmpdump: <a href="http://jeremy.visser.name/2009/08/30/python-iview/">python-iview</a> <a href="http://sourceforge.net/projects/mlbviewer/">mlbviewer</a> <a href="http://users.on.net/~m.elliott/i/index.html">iViewNapper</a> +<a href="http://mai9.net/wiki/index.php?title=TV3_al_Cab%C3%A0s">TV3 al Cabàs</a> Projects using librtmp:

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Compatibility with librtmp?
by Andrew McClure 16 Feb '11

16 Feb '11

Hi. We are developing an RTMP application and are having trouble getting it to interoperate with the librtmp (as used in FFmpeg). Our application acts like a proxy. In our test configuration, we have a port 1936 which accepts RTMP publishing, and a different port 1935 which accepts RTMP playing. Packets received from the publisher connected to port 1936 are forwarded to the player connected on port 1935 (with some minimal munging). We have two simple flash applets (written in Flex, using the standard Video/Camera/NetConnection objects) which connect to these ports; one streams video from a webcam, the other plays the video thus streamed. When we try to play video originating in Flex, forwarded through our proxy and played back in Flex, it works. The video appears on the screen. However then we try something different. We compiled a version of ffmpeg (svn r26326) with librtmp (from rtmpdump svn r555). (If there is a better way to test publish with librtmp than this, I would be curious to know.) We try using ffmpeg to publish to our port 1936: ./ffmpeg -loglevel debug -i ~/Downloads/20051210-w50s.flv -re -ac 1 -acodec copy -vcodec flv -f copy rtmp://localhost:1936/ And then play back the video by connecting the Flex applet to port 1935. We are having a problem where we are able to handshake, understand ffmpeg's publish commands and forward the packets through to the Flex applet as appropriate... *but* the Flex applet does not display video or audio. We get a black windowpane and silence. When we forward the ffmpeg streams, we appear to be doing (or failing to do) something to the RTMP packets such that the Flex player no longer understands the RTMP stream. But of course this problem only occurs with streams generated through ffmpeg. So it appears this is due to some slight difference between the RTMP protocol as implemented by ffmpeg and Flex-- we are compatible with the Flex publisher's notion of RTMP but not ffmpeg's. What I am trying to figure out: - What might this difference in RTMP implementation be, which is causing ffmpeg to apparently not be compatible with our RTMP proxy? - Is there a way to get ffmpeg (or another librtmp tool) to act like a "server", such that our Flex video object could connect directly to ffmpeg, issue a play request, and start receiving a video? I ask this in case the problem is not in fact that we are damaging ffmpeg's stream, but rather that ffmpeg just is not generating data Flex knows how to work with in the first place. (I find this "ffserver" tool, however this does not look like what I am looking for; the docs seem to imply it does RTP/RTSP only, and the html doc also says "It can also stream from files, though that is currently broken.") One possible source of concern is that when the player connects on 1935, it does not see the beginning of the stream-- it comes in after the stream has been waiting a few seconds. However once connected we do wait until we see a keyframe to begin streaming, and we forward a cached copy of the metadata packet at that time. We also did try forcing ffmpeg to publish in codecs I know Flex understands, but it does not help: ./ffmpeg -loglevel debug -i ~/Downloads/20051210-w50s.flv -re -ac 1 -acodec nellymoser -vcodec flv -f flv rtmp://localhost:1936/ Do you have any suggestions? What are the requirements our application must follow to interoperate with ffmpeg/librtmp? To our understanding we are compatible with the published RTMP spec but the spec is ambiguous on many points. Note, I was unsure the most appropriate place to ask these questions so I also sent this post to the ffmpeg-user list. Thanks.

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RTMPSockBuf_Send - Blocking
by Steve McFarlin 10 Feb '11

10 Feb '11

This message is in case someone runs across the same issue when using librtmp to publish on unreliable networks using iOS. Send will block indefinitely when one of the radios goes down (iphone). SIGPIPE may be thrown if there is no radio available, but this is inconsistent at best. The only way I have found around this issue is to use select with a timeout of 0. steve

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