[SR-Users] (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Alexandr Usov
blessendor at gmail.com
Tue Aug 6 11:02:35 CEST 2013
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 at 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 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
> > 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 at 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 at lists.sip-router.org
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>
>
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