13 Apr
2005
13 Apr
'05
7:27 a.m.
Gcc,
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.
SER offers two solutions namely rtpproxy and mediaproxy. These are complete
independent of each other and both have pros and cons.
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.
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.
Regards,
Paul
On 4/13/05, gcc chentob <chentob_gcc(a)hotmail.com> wrote:
Hello javarockx,
Thank you very much ! I can through nat by the SER+rtpproxy, but there
are some problem about SER I want to know. could you help me.
One is :when using SER +rtpproxy and all of audio data must through the
SER, Is it a good way?Do you test the SER it can support how many ways of
audio ?
The other is : Can the SER support video?
Thank you!
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