Hi Andreas,
Thank you very much. This is a good start for me. I also need to work with
freeswitch. Not asterisk. And simple point to point sip is more than enough
and rtp bridge would be fantastic.
Would it work with amr codec as well ? Or only gsm full rate/ h. Rate?
Im trying to find the lcr git location of jolly/new. Can you please give me
the url. I cant find it on openbsc git.
Thanks again.
Nava.
On Tuesday, October 16, 2012, Andreas Eversberg wrote:
hi kelum,
in order to use sip, you need jolly/new branch of lcr and jolly/rtpmux of
openbsc. the sip implementation of LCR is quite simple, so no
authentication or oder features - just simple point-to-point SIP. if you
run confiure of LCR, check if sip is enabled. in order to add a SIP
interface, do the following at interface.conf:
[sip]
sip <local ip>[:<local port>] <remote ip>[:<remote port>]
earlyb yes
tones no
you need to define local IP that will be used to connect to remote SIP
endpoint. don't use localhost, if the endpoint is on a different machine,
because this IP is also used for RTP. if you use same machine, you need to
have different ports. you may change local port, by adding a local port or
you may change port of SIP endpoint and then add remote port.
i have tested it with freeswitch, but asterisk should work also.
you may then also try at interface.conf below "[sip]" definition:
rtp-brige
then the codec (full rate or enhanced full rate) is negotiated between
mobile and the remote SIP endpoint. the SIP endpoint must support at least
one of it.
regards,
andreas