[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,

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.live555.com/pipermail/live-devel/attachments/20141112/e3d8806a/attachment-0001.html>


More information about the live-devel mailing list