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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Hi Manas, <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>My pointers are inlined-<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 color=blue face="Times New Roman"><span
style='font-size:12.0pt;color:blue'>Q1:I want to capture and encode both video
and audio and stream it to another end and receive RTP packets from the other
end and decode and play it. So whether I have to run two RSTP programmes
ie. both server and client ? . Can't I do this in a single programme.<br>
<br>
</span></font><font color=navy><span style='color:navy'><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Ans: Before deciding on whether both RTSP
server & client need to be implemented, you can once give a thought whether
RTSP will fit into video telephony scenario. SIP is certainly better choice.
How are you going to discover the IP address and port where the callee is
listening for request from a caller (in SIP term the ‘Contact’
header)? <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>However, for the time being if you have
decided to keep your callee and caller with a static address (and that is
reachable with respect to each other), you can go with RTSP. In that case, the caller
must act as RTSP client and the callee must act as RTSP server. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>But note that both of the callee and
caller must act as RTP client & server.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
Q2: I went through the test programme . I found that we can play only
mpeg1 audio and video together. Can't I use mpeg4 and mp2 audio together and
also stream them together.<br>
<br>
<font color=navy><span style='color:navy'><o:p></o:p></span></font></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Ans: Yes, you can. You have to write your
own program that will chose appropriate RTPSink and sources. Though I may be
wrong, but combination of mpeg4 and mp2 is quite non-standard. For speech,
there are much lightweight codec, isn’t? <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
<font color="#0080ff"><span style='color:#0080FF'>Q3: Is the live555 media
server takes care of packet losses in the network and synchronization between
the sender and receiver when both are sending in either directions. Or I
have to write my own code to do that.<br>
<br>
</span></font><font color=navy><span style='color:navy'><o:p></o:p></span></font></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Ans: To best of my knowledge, Live555
doesn’t have any exclusive mechanism to handle packet loss. The
synchronization mechanism is essentially limited to the fact that the server
tries to throw the packet in the network at appropriate time interval (or more
technically according to PTS).<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
Q4: The encoder put the encoded video and
audio data in two separate file and the decoder can take video and audio
data from two separate files. If this is my case can I implement the live555
programe for my purpose effortlessly as it can take the files as input and
output..<br>
<br>
<font color=navy><span style='color:navy'><o:p></o:p></span></font></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Ans: The file approach will degrade
performance for real time purpose. You can develop your own source classes,
which can directly take the encoded data and may store in some in-memory data-structure.
Then, the appropriate sink will form RTP packets using the encoded data stored
within that custom source.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Reverse will be the case in decoding side.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Regards,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Shaswata<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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