[SR-Users] (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Daniel-Constantin Mierla
miconda at gmail.com
Mon Aug 5 23:02:13 CEST 2013
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 at sip1.somedomain.com.ua
> <mailto:1-100 at sip1.somedomain.com.ua>
> -- SIP/sip1.somedomain.com.ua-000004cf is ringing
> -- SIP/sip1.somedomain.com.ua-000004cf answered SIP/1-101-000004ce
> > 0x15bc370 -- Probation passed - setting RTP source address to
> 1.1.1.1:50868 <http://1.1.1.1:50868>
> > 0x7f2b6044bd10 -- Probation passed - setting RTP source
> address to 1.1.1.1:35082 <http://1.1.1.1:35082>
>
> Got RTP packet from 1.1.1.1:50868 <http://1.1.1.1:50868> (type 00,
> seq 027109, ts 000160, len 000160)
> Sent RTP packet to 1.1.1.1:35082 <http://1.1.1.1:35082> (type 00, seq
> 037469, ts 000160, len 000160)
> Got RTP packet from 1.1.1.1:50868 <http://1.1.1.1:50868> (type 00,
> seq 027110, ts 000320, len 000160)
> Sent RTP packet to 1.1.1.1:35082 <http://1.1.1.1:35082> (type 00, seq
> 037470, ts 000320, len 000160)
> Got RTP packet from 1.1.1.1:50868 <http://1.1.1.1:50868> (type 00,
> seq 027111, ts 000480, len 000160)
> Sent RTP packet to 1.1.1.1:35082 <http://1.1.1.1:35082> (type 00, seq
> 037471, ts 000480, len 000160)
> Got RTP packet from 1.1.1.1:50868 <http://1.1.1.1:50868> (type 00,
> seq 027112, ts 000640, len 000160)
> Sent RTP packet to 1.1.1.1:35082 <http://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 list
> sr-users at lists.sip-router.org
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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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