Hi!
Thanks guys! I got it to work yesterday morning ... I have audio going through a chain of 2 rtpproxies ... My first attempts failed ... i mistook the F parameter by the R one ... :( I basically do as Atle showed on his email ... plus some unforce_rtp_proxy here and there (i found it in some onsip.org config file).
Cesc
On 12/11/06, Klaus Darilion klaus.mailinglists@pernau.at wrote:
Cesc wrote:
On 12/9/06, Atle Samuelsen clona@cyberhouse.no wrote:
Hi Cesc,
Thanks for VON.
It was my pleasure to meet all you guys ... I am hooked now. In my mind I have a background process trying to figure out how to go to the next one (San Diego? :D )
I have a few questions ...
- i saw mentioned that chaining is possible ... no problem there
right? i need to send an extra parameter to the force_rtp_proxy and that is it? no side-effects if, i.e, call between the phones in the same island (thus, just one rtp proxy)?
This is possible, but you need to turn of the checking :) (there is a modparam that checkes if there in the sdp sasys a=nortpproxy or something like that.
Ok ... i think is a parameter in force_rtp_proxy, right?
yes, you need the "f" flag (guess it is still the same in ser)
http://www.openser.org/docs/modules/1.1.x/nathelper#AEN275
A final question ... basically thinking out loud (and writing it down) :) I read that rtpproxy won't start relaying until it got an rtp packet from both sides ... is it true? could this not cause problems, specially with chained rtpproxies, if say, i have one of the phones not sending rtp packets (say, it starts muted ... muted means no rtp packets)?
AFAIR rtpproxy is asynchronous until the first RTP packet from each side is received. Thus chaining should work.
regards klaus
btw: rtpproxy also has some parameters: http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/rtpproxy/manpage.xml?rev=1.2&a...
also main.c wil show you that there are some more undocumented parameters.
Cesc
- Atle
Regards,
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