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
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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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?
Cheers, Mo
Date: Thu, 17 Jul 2014 13:56:53 +0200 From: miconda@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 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
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@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 sr-users@lists.sip-router.org <mailto: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://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
Hi,
Is it possible to create a static user/extension in Kamailio? I.e.: Have extension 11223344 being sent to IP address 192.168.0.100 while also not requiring this extension to authenticate when making calls.
Cheers! Moacir
On Friday 18 July 2014 16:31:08 Moacir Ferreira wrote:
Is it possible to create a static user/extension in Kamailio? I.e.: Have extension 11223344 being sent to IP address 192.168.0.100 while also not requiring this extension to authenticate when making calls.
Setting $rd to "192.168.0.100" should do the trick if $rU equals "11223344". It is up to the endpoint whether to accept this call or not.
Thanks for the deedback Daniel. I thought there could be an easier way to do it as we may have several devices like this. But as I got no other responses to my query, then I guess this is the only way to do it!
Cheers! Moacir
From: daniel@pocos.nl To: sr-users@lists.sip-router.org Date: Mon, 21 Jul 2014 11:30:21 +0200 Subject: Re: [SR-Users] Non authenticated user/extension
On Friday 18 July 2014 16:31:08 Moacir Ferreira wrote:
Is it possible to create a static user/extension in Kamailio? I.e.: Have extension 11223344 being sent to IP address 192.168.0.100 while also not requiring this extension to authenticate when making calls.
Setting $rd to "192.168.0.100" should do the trick if $rU equals "11223344". It is up to the endpoint whether to accept this call or not.
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
On Monday 21 July 2014 19:48:10 Moacir Ferreira wrote:
Thanks for the deedback Daniel. I thought there could be an easier way to do it as we may have several devices like this. But as I got no other responses to my query, then I guess this is the only way to do it!
Kamailio has to know about these devices somehow, since they aren't registering you either have to hardcode or use alias_db_lookup (from alias_db module) or some other custom db queries.
Hi there!
What are the benefits of using the new mediaproxy-ng over the old rtpproxy?
Thanks, Moacir
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@gmail.com To: sr-users@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@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 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
_______________________________________________ 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
I am running Kamailio 4.1.4 and rtpproxy 1.2.0 on a dual-homed server. In another server I have a Linux video MCU working on the private LAN. Everything works quite nice from the private and Internet even if the client is behind a firewall.
However, I decided to host the MCU on the same server that I have Kamailio, changing the SIP port of the MCU to 5070. WHile the MCU is still functional, the "system" just stop working… Kamailio is fully functional for phone-to-phone calls, including video. The MCU is fully functional if you call it directly. However, if I call the MCU via Kamailio, that is on the same server, I will never get anything on the client from the MCU. The softphone will send the RTP to the server but I am not getting the traffic from the MCU arriving at the softphone.
Am I missing something? Any hint on what I should care in this kind of environment?
Cheers!
Moacir
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Daniel Tryba -
Daniel-Constantin Mierla -
Moacir Ferreira -
Pascal Fautré