On 17/07/14 23:10, Moacir Ferreira wrote:
I have created an environment with the same config and I find the same
problem. While still does not work for video, I have changed (flip)
the public/internal IP addresses on rtpproxy and I can get half call
leg working properly, includding video.
However, I am testing video calls. So I got another question on top of
the original post: Can we use rtpproxy also for video or it only
supports voice rtp proxy?
Yes, it works for both audio and video at the same time. As an example,
see my ipv4-ipv6 tutorial where I used it in bridge mode and tested with
video using Jitsi:
- http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
Cheers,
Daniel
Cheers,
Mo
------------------------------------------------------------------------
Date: Thu, 17 Jul 2014 13:56:53 +0200
From: mico...@gmail.com
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Kamailio RtpProxy MHomed
Hello,
have you looked at sip trace and checked what are the IP addresses in
the SDP? Maybe you need to swap the flags i and e.
You can eventually provide here the incoming invite as well as
outgoing invite, saying what you would expect to be in the outgoing
one, so we can give further hints.
Cheers,
Daniel
On 16/07/14 15:08, Pascal Fautré wrote:
Hi,
I tried to use Kamailio / RTPProxy in mhomed setup without any luck.
I had no problem to configure it with only 1 interface, without
mhomed, everything worked perfectly.
The RTP streams where not established correctly even if I managed
to have to proper IP in the SIP INVITE (C & O).
Versions:
version: kamailio 4.1.4 (x86_64/linux)
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP,
PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST,
HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN
16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 04:23:19 Jun 13 2014 with gcc 4.7.2
RTPProxy -v:
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the
update/lookup command
Extension 20081224: Support for session timeout notifications
Here is my RTPProxy config (/etc/default/rtpproxy) :
CONTROL_SOCK=udp:127.0.0.1:7722
EXTRA_OPTS=“-l /PU.BL.IC.IP///PRI.VA.TE.IP/ -m 11000 -M 12000 -d
DBUG:LOG_LOCAL3
Here are snippets of my kamailio.cfg:
port=5060
mhomed=1
# 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;
xlog("NATMANAGE M=$rm OU=$ou RURI=$ru RD=$rd F=$fu T=$tu
NH=$nh(d) IP=$si ID=$ci\n");
if(dst_ip == PUBLIC_IP) {
if(is_ipv4($nh(d)) && is_in_subnet($nh(d), PRIVATE_NET)) {
xlog("NATMANAGE coei\n");
rtpproxy_manage("coei", PRIVATE_IP);
} else {
xlog("NATMANAGE coee\n");
rtpproxy_manage("coee", PUBLIC_IP);
}
} else {
if(is_ipv4($nh(d)) && is_in_subnet($nh(d), PRIVATE_NET)) {
xlog("NATMANAGE coii\n");
rtpproxy_manage("coii", PRIVATE_IP);
} else {
xlog("NATMANAGE coie\n");
rtpproxy_manage("coie", PUBLIC_IP);
}
}
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
if(is_first_hop())
set_contact_alias();
}
}
#!endif
return;
}
Calls were correctly going to the desired rtpproxy_manage options.
Now I’m not quite sure I’m using the correct ones.
I had to specify the PUBLIC_IP or PRIVATE_IP in the
rtpproxy_manage calls in order to have the correct IP address in
the C and O headers of the SIP INVITE. Without that, the public IP
would be sent as C and O params to phones on the private subnet.
In fact not a single call direction would give correct RTP streams.
Any idea where I missed the turn?
Cheers
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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--
Daniel-Constantin Mierla -http://www.asipto.com
http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda>
-http://www.linkedin.com/in/miconda
_______________________________________________ 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
_______________________________________________
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
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
_______________________________________________
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
