[SR-Users] Kamailio RtpProxy MHomed
Moacir Ferreira
moacirferreira at hotmail.com
Sat Jul 19 13:07:45 CEST 2014
I was using an Internet access from Vodafone that has a modem with a SIP ALG for their phone. Not sure why, this modem would prevent to connect properly. But Kamailio/rtpproxy was doing what it was supposed to do as it works on a modem with no ALG.
To fix the posted configuration problem, just flip the internal/external IP when starting rtpproxy. I think the information on how to start it is missleading.
Finally, I tested:
Internet <-> Internet. It works and rtpproxy is not used as supposed to be.
Internet <-> NATed device arriving at the external interface. It works and here rtpproxy is used as it is suppoed to be.
Internet <-> NATed device behind Kamailio (internal interface). It works and here rtpproxy is used as it is suppoed to be.
The only remaining test I want to do is between two devices, registered on internal network, calling eah other. Here rtpproxy should not be used as they are in the same subnet. However, I am afraid it will be used as the check for NATed devices will always be set as NATed if the call is comming from private address space (RFC1918). Am I wrong?
Cheers!
Moacir
Date: Fri, 18 Jul 2014 00:18:27 +0200
From: miconda at gmail.com
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] Kamailio RtpProxy MHomed
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: miconda at gmail.com
To: sr-users at 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
sr-users at 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 at 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 at 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 at lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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