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 gminghon@gmail.com:

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                                       |                                       v                               [ASTERISK] what i need to achieve is UA1 call UA2 and rtp traffic flow to kamailio but not to asterisk. can kamailio handle the rtp traffic it own? [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2]                                      ^                                      |                                      X                                      |                                      v                              [ASTERISK]

Thanks in advance.

Regards,

MingHon

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 should see reINVITE messages from Asterisk to the clients. Take a look at the SDPs in those requests and their responses to find out if they are malformed.

regards Klaus

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.

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. 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 the reinvite work. both ua will have and direct media flow

[ua1]<====>[ua2] | | x | v [asterisk]

when ua register to kamailio the audio work and the reinvite message is same as the first invite message.

[ua1]<====>[kamailio]<====>[ua2] | ^ | | | | v | [asterisk]

how do i stop the media flow between kamailio and asterisk? make kamailio relay the rtp between both ua.

[ua1]<====>[kamailio]<====>[ua2] | ^ x x | | v | [asterisk]

anyone could give some hint?

thanks in adv.

-- Regards,

MingHon

---

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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 "" "" udp'

to capture all UDP packets and watch if RTP packets are sent by the phones and proper forwarded by rtpproxy.

regards Klaus

Am 21.07.2011 10:31, schrieb MingHon:

Hi klaus,

here is my ngrep i paste it pastebin

pls take a look.

http://pastebin.com/eHtMXvEx

thanks in adv.

Regards,

MingHon