similar to: RTSP reflection.

Atul S Vasu wrote: > Where do I find instruction to configure icecast as a reflect server for > rtsp? Icecast doesn't suppport RTSP. Only HTTP. Geoff.

Greetings: It would be nice if Icecast supported RTSP; however I would appreciate any suggestions for a small RTSP/RTP solution to encode 8kHz mono audio in GSM or ADPCM and service multiple unicast client connections. The ideal would be a black-box hardware solution with an audio input and ethernet interface similar to broadcast studio IP audio links or the network audio capabilities of certain

hi I am getting this error: -- Executing [50 at smvoice-sip:1] Answer("SIP/440-0856dd70", "") in new stack -- Executing [50 at smvoice-sip:2] rtsp("SIP/440-0856dd70", "rtsp://192.168.1.175/img/video.sav") in new stack [Apr 30 11:22:48] WARNING[8031]: app_rtsp.c:1037 rtsp_play: >rtsp play [Apr 30 11:22:48] NOTICE[8031]: rtp.c:1287 ast_rtp_read:

Hello, I am trying to stream an IP Camera output (h264) into a conference. The IP Camera supports RTSP. Searching around the web, I believe the RTSP support (was) available through app_rtsp (external to Asterisk distribution). This, I believe, has problems and has issues compiling in Asterisk 11 (I tried compiling it in Asterisk 11 and it failed). I may not be able to use DiaStar or i6net's

Thomas B. Ruecker wrote: > Michael Grigoni wrote: > >>Greetings: >> >>It would be nice if Icecast supported RTSP; > > It probably never will > >>however I would >>appreciate any suggestions for a small RTSP/RTP solution to >>encode 8kHz mono audio in GSM or ADPCM and service multiple >>unicast client connections. > > why not use

Hi I need to stream ogg/theora over RTSP in order to use in opera browser HTML 5 video, but I cannot find any streaming server to do that. -- best regards, Daniel Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/theora/attachments/20100311/f4ec72d2/attachment.htm

Dear Opus developers. This is the first time I write here, so hello to everybody! Sorry to disturb you but I would like to ask you something I could not answer by googling and by looking at this mailing list archive. I have just started investigating this new and promising codec for real-time audio transport over the internet for industrial applications. I was previously experimenting with RTP

I am having problems making RTSP connections to a Windows Streaming Media Server (ie "connecting to media...." but WMP never connects). There are no error messages in /var/log/messages. It was suggested to me that SNAT might perform better than MASQ in this respect. I edited my shorewall/masq file as such: eth0 eth1 12.34.56.78 or should it be? eth0 10.0.0.0/24

Hello All, I''m running performance tests on a Linux router using IPTables to nat traffic over the network. I have a MS Media streamer, and two windows clients behind the router which download video from the streamer. I can then measure performance. MS Media player rolls amongst protocols until it finds one it can use as follows RTSP UDP RTSP TCP MMS UDP MMS TCP HTTP Unfortunately it

I am trying to add a package (I know its older - but it should work) gst-rtsp-server-0.10.8 to CentOS 7. The gstreamer 0.10 packages are included in C7 and they compile just fine. When I extract and try to compile gst-rtsp-server 0.10.8 the ./configure goes fine. but the make results in errors: make make all-recursive make[1]: Entering directory

Hi, I have been trying to use wine to get at a streaming quicktime server on Ubuntu 8.04, but as of yet no luck: The server streams mov files over RTSP First I tried using just VLC to play it, but there are some things it just can't handle (in windows I get an error about 'sowl' not being supported or something like that) Now I am trying quick time alternative in wine along with

I am playing with gstreamer on CentOS 7.1 . All the packages are there except gst-rtsp-server.... OK - So I went to look for it to compile it. Found a version that was NOT 1.0.7 and my compile stopped because it said the versions are not the same.... OK so went back to look for 1.0.7 and its not there. I looked at: http://gstreamer.freedesktop.org/src/gst-rtsp-server Any thoughts on how I

On Thu, 2020-09-24 at 16:57 +0200, Antony Stone wrote: > On Thursday 24 September 2020 at 16:31:33, hw wrote: > > > Hi, > > > > is it possible to "call" an IP camera? > > Only if it talks SIP (which some do, generally door entry cameras with a > push > button input and often a lock release output). > > > I'm thinking about something

Thanks Arthur, > On 7/31/05, Richard <rpm@16hd.freeserve.co.uk> wrote: >>Can anyone advise whether there are any plans for icecast to support >>rtsp (real-time streaming protocol), as used by RealPlayer, Apple >>Quicktime etc.? >>Alternatively - though I really like icecast and want to keep with it >>perhaps someone can suggest an open-source

Hi, does anybody know if somebody tried implementing things like rtsp, mms or similar in icecast so far? I heard some rumors from people complaining transmissions over http or not "that secure" because using a wget you can easily write them to disk. They suggested using real servers or something - but I wonder if its (easy?) possible to implement another already widely used

I have a gst-rtsp-server 0.10.8.2 that I am trying to compile on CentOS 7.2 I get this error: rtsp-media.c error 'g_cond_free' is deprecated (declared at /usr/include/glib-2.0/glib/deprecated/gthread.h:283) [-Werror=deprecated-declarations] What can I do about that to compile this package (I'm compiing from source) ? Thanks Jerry

On 19 Jun 2003 at 21:20, Arc wrote: > On Thu, Jun 19, 2003 at 09:06:34PM -0400, Steve Lewis wrote: > > > > Right now I just have a link to an MP3 file and it does HTTP > > transfers of the music to whoever clicks on it. It is sometimes very > > slow and that's why they have offered to put in a streaming server. > > Do I just have the plain link to the MP3

Hi, I'm building an rtsp client/server that will support vorbis and I'm trying to determine the best way to play the vorbis content as the array is continuously being filled. Currently, I can download an ogg file via rtsp and write it out to a FILE and then play_file (from ogg123) it when it's done -- so the file is in the correct format. What would you recommend? -- Wayne Davis

Are there any plans to provide support to stream to cellphones ? If my understanding is correct, this requires 3Gpp, RTP/RTSP support. I see that apple has a stream server for this. Randy

On Friday 20 June 2003 11:37, Dave St John wrote: > isnt RTSP Protocol real networks genius?? > anyone?? Real uses it, and were involved with designing it, but it's a publically documented protocol (there's an RFC). It's a sort of weird-extended-stateful-HTTP-hybrid. Mike --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: