Re: [Live-devel] Error in compiling LiveMedia

I am trying to compile live.2008.05.12.tar.gz on QNX 6.2 system and I don't have watcom compiler, so I have replaced cc32 with gcc and during compilation, I am getting lot of syntax error and other errors like /include/ByteStreamF

[Live-devel] Error in compiling LiveMedia

Hi, I am trying to compile live.2008.05.12.tar.gz http://www.live555.com/liveMedia/public/live.2008.05.12.tar.gz > on QNX 6.2 system and I don't have watcom compiler, so I have replaced cc32 with gcc and during compilation, I am getting lot of syntax error and other errors like /include/ByteSt

Re: [Live-devel] Live555 jpeg streaming (1 pic per sec)

How can i use the live555 to stream one jpeg per 1 second. I had tried this with a thread and this thread wait 1 sec everytime but in Wireshark i see that the jpegs are sending every 4 sec. Which parameters in the Live555 and how can i set it for realizing that? First, if you haven't alrea

Re: [Live-devel] Creating an RTP Translator

On Thu, May 29, 2008 at 4:57 PM, Ross Finlayson <[EMAIL PROTECTED]> wrote: You're correct that you can't use the usual *RTP*Source and *RTP*Sink classes. However, because you wish to pass the RTP header fields 'as is', you could instead read and send raw-UDP (rather than RTP) packets. You ca

[Live-devel] Live555 jpeg streaming (1 pic per sec)

Hi, How can i use the live555 to stream one jpeg per 1 second. I had tried this with a thread and this thread wait 1 sec everytime but in Wireshark i see that the jpegs are sending every 4 sec. Which parameters in the Live555 and how can i  set  it for realizing  that? thank you Armando _

Re: [Live-devel] Creating an RTP Translator

On Thu, May 29, 2008 at 4:57 PM, Ross Finlayson <[EMAIL PROTECTED]> wrote: > You're correct that you can't use the usual *RTP*Source and *RTP*Sink > classes. However, because you wish to pass the RTP header fields 'as is', > you could instead read and send raw-UDP (rather than RTP) packets. You c

Re: [Live-devel] Creating an RTP Translator

I would like to use live555 to create and RTP translator that simply encrypts the payload of the RTP packets and preserves header information such as SSRC as required by the RTP standard. It seems to me I cannot use the usual source/sink facilities since then the sink would create new header infor

[Live-devel] Creating an RTP Translator

Hello, I would like to use live555 to create and RTP translator that simply encrypts the payload of the RTP packets and preserves header information such as SSRC as required by the RTP standard. It seems to me I cannot use the usual source/sink facilities since then the sink would create new heade