That message was apparently spam; you should not click on the link
that was inside it.
The email address that sent that message has now been unsubscribed
from the mailing list.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
___
live-de
Title: Yoomba
I am calling you on this great new thing called Yoomba.
Yoomba allows you to call and chat with ANYONE, simply using your email.
You can call me back, and we can talk for free!
Click here to call back
Let's talk, let's start the Yo
Hey all,
I am trying to send live audio from a microphone through Media Server to my
client. Any ideas?
Thanks
Jimmy
___
live-devel mailing list
live-devel@lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
>Ok, but I still don't get how the final recipient gets to know the
>*original* presentation time (which correspond to audio) if I generate a
>new one in the transcoder...
Each incoming frame (from your "RTPSource" object) has an accurate
presentation time, which you know (because it's passed as
>Does anyone have any idea why this could happen? Why is my RTP sink
>generating bad sequence numbers? What about the RTP packets with
>payload 72! They must be coming from the same RTPSink
No. Unless you have modified the existing library code, your
"alternating sequence numbers" must be comi
> Again, it sounds like you're trying to reinvent the wheel. The
> "OnDemandServerMediaSubsession" class works just fine - you should
> just use it (by defining your own subclass). Note the several
> examples in the code. You should be using "testOnDemandRTSPServer" -
> not "testMPEG4VideoStream
Thanks for the answer Ross. I will make myself a bit more clear to state the
case.
I am streaming a sequence of PES (both audio and video). I use the demux
class to stream audio and video separatelly ( audio to IP port and
video to ). I am using a dedicated network - so all traffic to th
Again, it sounds like you're trying to reinvent the wheel. The
"OnDemandServerMediaSubsession" class works just fine - you should
just use it (by defining your own subclass). Note the several
examples in the code. You should be using "testOnDemandRTSPServer" -
not "testMPEG4VideoStreamer" -
Sorry, I forgot ask you about SDP file. Now address is correct (0.0.0.0) but
what about port? "m" line of SDP is still using port I specified when I create
groupsock, could I change it? Should I change it? If the first description is
sending broadcast in te same port it will cause a problem.
Thank
> You probably shouldn't be using "addDestination()" - that is a
> specialized function used only to implement on-demend unicast
> streaming to multiple clients from a single source. (Note that, for
> unicast on-demand streams, the SDP description should contain the
> special address 0.0.0.0, not
You probably shouldn't be using "addDestination()" - that is a
specialized function used only to implement on-demend unicast
streaming to multiple clients from a single source. (Note that, for
unicast on-demand streams, the SDP description should contain the
special address 0.0.0.0, not a spec
Il giorno mer, 11/07/2007 alle 10.19 -0700, Ross Finlayson ha scritto:
> I did some more testing on this "rtsp://" URL, and the problem
> definitely seems to be at the server end. For some reason, your
> server is not responding properly (if at all) to RTSP requests.
Thanks for your help.
I wil
Hi all,
I created a session with its respective rtpGroupsock & rtcpGroupsock and then
I added another destination using:
rtpPort= new Port(newPort);
rtcpPort= new Port(newPort+1);
destinationAddress.s_addr = inet_addr((char*)newIP);
rtpGroupsock->addDestination(destinationAddress,*rtpPort);
rtcpGr
13 matches
Mail list logo
