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