Gcc,<br>
<br>
When dealing with NATed SIP clients you have to handle their RTP (aka,
audio and/or video) in a special manner. The reason is that the two SIP
UAs in the call cannot directly talk to each other.<br>
<br>
SER offers two solutions namely rtpproxy and mediaproxy. These are
complete independent of each other and both have pros and cons.<br>
<br>
You can run either of these proxies on physically different servers
from SER. So to answer your first question, yes this is a good way.
It's not necessarily a good thing to run these media proxies on the
same physical server as SER, but nonetheless it's better than no audio.<br>
<br>
To answer your question about video, the answer, AFAIK, is yes. Both
rtpproxy and mediaproxy support streaming video. I've never tried
video, but there should be nothing special to do. If you search the
serusers mailing list archives you'll likely find other postings
related to video.<br>
<br>
Regards,<br>
Paul<br><br><div><span class="gmail_quote">On 4/13/05, <b class="gmail_sendername">gcc chentob</b> <<a href="mailto:chentob_gcc@hotmail.com">chentob_gcc@hotmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello javarockx,<br> Thank you very much ! I can through nat by the SER+rtpproxy, but there<br>are some problem about SER I want to know. could you help me.<br> One is :when using SER +rtpproxy and all of audio data must through the
<br>SER, Is it a good way?Do you test the SER it can support how many ways of<br>audio ?<br> The other is : Can the SER support video?<br> Thank you!<br><br>_________________________________________________________________
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