Hi,
On Thu, 2011-07-21 at 15:23 +0800, MingHon wrote:
> Hello List,
>
>
> im still trying but no luck.
> asterisk canreinvite already set to yes
>
What version of asterisk? I think in 1.6.2 canreinvite was replaced
with directmedia and directrtp.
>
> now im testing in lan
> i setup kamailio
Actually the trace looks fine.
You have to debug on network level: check if RTP packets are sent. Your
scenario looks like:
Caller rtpproxy(a)---rtpproxy(b)callee.
You have 2 instances of rtpproxy activated - which should send RTP
packets to each other.
e.g. use
'ngrep -d any -t -q -P
anyone??
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Hi klaus,
here is my ngrep i paste it pastebin
pls take a look.
http://pastebin.com/eHtMXvEx
thanks in adv.
--
Regards,
MingHon
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You should post a SIP trace, together with the IP addresses of all nodes:
ngrep -t -d any -P "" -Wbyline port 5060
If there is sensitive information in the traces, just remove/replace it.
regards
Klaus
Am 21.07.2011 09:23, schrieb MingHon:
> Hello List,
>
> im still trying but no luck.
> aster
Hello List,
im still trying but no luck.
asterisk canreinvite already set to yes
now im testing in lan
i setup kamailio and asterisk in same lan
kamailio&rtpproxy on 192.168.2.3 and asterisk on 192.168.2.23
canreinvite=yes in asterisk. when both ua in the same lan
register directly to asterisk t
Hi. Kamailio and rtpproxy does NOT decide to send rtp to asterisk. It is
asterisk who decides to receive it and that entirely depends on asterisk
sip condigurarion and asterisk sip peers configuration.
Your question is not related to kamailio but just to asterisk.
Hi,
after asterisk reinvite i get status 491: request pending.
after few seconds i hang up both UA then one of the UA will start ring.
please advice.
--
Thanks and Regards,
MingHon
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Hi,
yup i tried "canreinivte=yes" in sip.conf and also in the extension
database.
urm how bout having direct rtp traffic and also relay rtp traffic in my
setup?
example, UA1 and UA2 is at the same nat. so UA1 and UA2 will have direct rtp
traffic.
UA1 <--(rtp)--> UA2.
and UA3 and UA4 both behin
Am 13.07.2011 10:07, schrieb MingHon:
> Hi,
>
> i tried set canreinvite=yes in asterisk but kamailio/rtpproxy still
> trying to send rtp traffic to asterisk.
That should not happen. You have to investigate why. You have to take a
look at the SIP signaling during and after call setup.
You shoul
Hi,
have you tried "canreinvite=yes" on your Asterisk-box?
If that does not help, there is probably no way to make the
RTP-Traffic bypass your asterisk box...
Carsten
2011/7/14 MingHon :
> Hello,
> anyone?
> currently my setup look like this.
> when UA1 call UA2 and the rtp traffic flow to kama
Hello,
anyone?
currently my setup look like this.
when UA1 call UA2 and the rtp traffic flow to kamailio and asterisk.
[UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2]
^
|
RTP TRAFFIC
Hi,
i tried set canreinvite=yes in asterisk but kamailio/rtpproxy still trying
to send rtp traffic to asterisk.
and asterisk did not forward the rtp traffic back to kamailio/rtpproxy then
i will get no audio on the ua.
please adv.
thanks,
Regards,
MingHon
_
I guess you forward all calls via Asterisk.
Yes: set canreinvite=yes (name was changed in newer Asterisk versions)
in sip.conf for the peers and Asterisk will send reINVITEs after call
setup to offload RTP.
regards
Klaus
Am 13.07.2011 07:43, schrieb MingHon:
> Hi List,
>
> i would like to know
Hi List,
i would like to know is it possible to bypass the rtp traffic forwarding to
asterisk server?
my kamailio and rtpproxy is on the same box and asterisk is on the other
box.
can kamailio/rtpproxy handle the rtp traffic without forwarding to asterisk
box?
thanks in advance.
--
Regards,
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