[Live-devel] Streaming to RTSP clients with different ports
Taha Ansari
taha.ansari at objectsynergy.com
Wed Nov 12 02:48:06 PST 2014
Hi,
I have a test scenario, where I am streaming webcam input + mic over to a
remote server using RTP protocol, which re-streams this to another client
(other client pulls data through RTSP):
FFmpeg (RTP AAC audio + RTP video) -> Server -> (RTSP AAC audio + RTSP
video) FFmpeg client
In a local development setup, everything works fine. But in live
environment, neither the audio nor video gets pulled by other client. I
tried to use RTSP tunneled over HTTP, and that got my video content on other
side, but audio still fails.
If I enable debug level log in FFmpeg, it gives me this error:
method PLAY failed: 500 SERVER ERROR
and later:
x-Error: Failed to create audio
This error is given for both local and live server environment, so I believe
this could be a server related fault (remember this error is there only for
AAC, video contents are delivered/fetched just fine using HTTP tunneling).
Being very new to live555, I am not sure how to trigger a command line based
live555 instance that will read an input URL (through SDP maybe), and
broadcast arriving data over RTSP on some port; if anyone here can guide me,
I will really appreciate it.
For reference, this is complete console output when fetching stream from
live server:
------------------------------------------------------------------
ffplay -loglevel debug -rtsp_transport http
rtsp://171.215.211.115:3648/a.aac
ffplay version N-63439-g96470ca Copyright (c) 2003-2014 the FFmpeg
developers
built on May 25 2014 22:05:32 with gcc 4.8.2 (GCC)
configuration: --disable-static --enable-shared --enable-gpl
--enable-version3
--disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig
--ena
ble-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray
--e
nable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm
--enable-libi
lbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb
--enable-
libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp
--enabl
e-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora
--enable
-libtwolame --enable-libvidstab --enable-libvo-aacenc
--enable-libvo-amrwbenc --
enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp
--enable-l
ibx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink
--en
able-zlib
libavutil 52. 86.100 / 52. 86.100
libavcodec 55. 65.100 / 55. 65.100
libavformat 55. 41.100 / 55. 41.100
libavdevice 55. 13.101 / 55. 13.101
libavfilter 4. 5.100 / 4. 5.100
libswscale 2. 6.100 / 2. 6.100
libswresample 0. 19.100 / 0. 19.100
libpostproc 52. 3.100 / 52. 3.100
[http @ 022a46a0] request: GET /a.aac HTTP/1.10KB sq= 0B f=0/0
User-Agent: Lavf/55.41.100
Range: bytes=0-
Connection: close
Host: 171.215.211.115:3648
x-sessioncookie: bceb6136e1a59ada
Accept: application/x-rtsp-tunnelled
Pragma: no-cache
Cache-Control: no-cache
[http @ 022a46a0] header='HTTP/1.1 200 OK' 0KB sq= 0B f=0/0
[http @ 022a46a0] http_code=200
[http @ 022a46a0] header='Date: Wed, 12 Nov 2014 04:59:44 GMT'
[http @ 022a46a0] header='Server: [server name here]'
[http @ 022a46a0] header='Connection: Close'
[http @ 022a46a0] header='Content-Type: application/x-rtsp-tunnelled'
[http @ 022a46a0] header='Expires: -1'
[http @ 022a46a0] header='Cache-Control: private, max-age=0'
[http @ 022a46a0] header=''
[http @ 022a4780] request: POST /a.aac HTTP/1.1KB sq= 0B f=0/0
User-Agent: Lavf/55.41.100
Accept: */*
Connection: close
Host: 171.215.211.115:3648
x-sessioncookie: bceb6136e1a59ada
Content-Type: application/x-rtsp-tunnelled
Pragma: no-cache
Cache-Control: no-cache
Content-Length: 32767
Expires: Sun, 9 Jan 1972 00:00:00 GMT
[rtsp @ 022a4ae0] SDP:= 0 aq= 0KB vq= 0KB sq= 0B f=0/0
v=0
o=- 400522656 1415768385 IN IP4 171.215.211.115
s=
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 96
a=rtpmap:96 MPEG4-GENERIC/22050/2
a=fmtp:96
profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdelta
length=3;config=1390;
a=control:trackID=2
[rtsp @ 022a4ae0] audio codec set to: aac
[rtsp @ 022a4ae0] audio samplerate set to: 22050
[rtsp @ 022a4ae0] audio channels set to: 2
[rtsp @ 022a4ae0] hello state=0 0KB vq= 0KB sq= 0B f=0/0
[rtsp @ 022a4ae0] method PLAY failed: 500 SERVER ERROR 0B f=0/0
[rtsp @ 022a4ae0] Server: [server name here]
CSeq: 4
Cache-Control: no-cache
Date: Wed, 12 Nov 2014 04:59:46 GMT
Expires: Wed, 12 Nov 2014 04:59:46 GMT
Session: 57538100428063;timeout=30
x-Error: Failed to create audio
rtsp://171.215.211.115:3648/a.aac: Invalid data found when processing input
------------------------------------------------------------------
Note: some info changed about to protect privacy.
Thanks in advance for all guidance!
Regards,
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