6 Aug
2013
6 Aug
'13
5:14 a.m.
Dear Alexandr,
You can connect Kamailio to RTPproxy via socket as well, use modparam like
this:
modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:12221")
Then if your rtprpoxy is started in bridged mode you should use the "i" and
"e" flags while you call the rtpproxy-manage() function in the kamailio.cfg
file.
The placement of both the flags sets the SDP c= param , so if you use "ie"
combination of flag then that is not equal to "ei" combination of the flag.
I also suggest that you turn on sip debug on the call receiving asterisk
and observe the SDP for an incoming call from Kamailio. that will help you
figure out the situation in SDP.
Best Regards,
Sammy
On Tue, Aug 6, 2013 at 2:02 AM, Alexandr Usov <blessendor(a)gmail.com> wrote:
Thank you for response!
A little difficult for me to find the same logic in my case with tutorial
of ipv4/ipv6 bridgin...
When I started
/usr/sbin/rtpproxy -u kamailio -l 1.1.1.1/2.2.2.2 -s udp:127.0.0.1 12221
There is no sound.
Is this a major to connect via unix sock?:
modparam("rtpproxy", "rtpproxy_sock",
"unix:/var/run/rtpproxy/rtpproxy.sock")
2013/8/6 Daniel-Constantin Mierla <miconda(a)gmail.com>
Hello,
you have to use rtpproxy in bridge mode, to route packets between the two
local network interfaces. There are many examples out there, one shows even
how to bridge between ipv4 and ipv4 networks -- you can use it as reference:
- http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
Cheers,
Daniel
On 8/5/13 7:12 PM, Alexandr Usov wrote:
I have Kamailio on OpenSUSE with static real Public IP (WAN), for ex.
1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM virtual machine with LAN IP
2.2.2.101 and default GW not the SuSe (2.2.2.2), but 2.2.2.1 pfsense LAN
with PUB IP 1.1.1.2) I am configured Registration of UA on Kamailio DB, and
on Asterisk side create a static peers with Kamailio LAN ip (host=2.2.2.2).
RTP Proxy question.
/usr/sbin/rtpproxy -u daemon -l 1.1.1.1 -s udp:127.0.0.1 12221
.... kamailio.cfg ...
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
Testing call:
Whe User 1-100 calling User 1-101, on Asterisk side I see:
-- Called SIP/1-100(a)sip1.somedomain.com.ua
-- SIP/sip1.somedomain.com.ua-000004cf is ringing
-- SIP/sip1.somedomain.com.ua-000004cf answered SIP/1-101-000004ce
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
0x15bc370 -- Probation passed - setting RTP
source address to
1.1.1.1:50868
0x7f2b6044bd10 -- Probation passed - setting RTP
source address
to 1.1.1.1:35082
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027109, ts 000160,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037469, ts 000160,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027110, ts 000320,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037470, ts 000320,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027111, ts 000480,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037471, ts 000480,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027112, ts 000640,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037472, ts 000640,
len 000160)
Voice transfers OK.
But why not Kamailio LAN ip I receiving on the Asterisk side with the
same LAN?
And Kamailio log grep:
skynet:~ # tail -f /var/log/messages | grep rtpproxy
2013-08-05T19:18:17.508760+03:00 skynet kamailio[25462]: 3(25481) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.508875+03:00 skynet kamailio[25462]: 3(25481) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1
2013-08-05T19:18:17.530765+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.530876+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 41958 1.1.1.1
2013-08-05T19:18:18.625815+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.627131+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 39876 1.1.1.1
2013-08-05T19:18:18.632649+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.648075+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.649615+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1
2013-08-05T19:18:18.653948+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.688603+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.689762+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1
2013-08-05T19:18:18.701062+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.701405+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1
2013-08-05T19:18:18.705506+03:00 skynet kamailio[25462]: a=nortpproxy:yes
My goal is using Asterisk boxes behind Kamailio with the same LAN or
even OpeVPN Lan2Lan, with Public IP on Kamailio WAN for users registration
and RTP routing. So is strange to my, why RTPproxy not rewrite source of
RTP traffic from PUBLIC Kamailio IP to LAN Kamailio IP when user A calls B
via Asterisk?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users