At 13.51 19/12/2006, Klaus Darilion wrote:
You said that the 200 contains openser's IP in the SDP? Is it put in
there by openser or already by Asterisk?
Tnx very much for help
It's put in there by OpenSER.
I'm attaching the 2 SIP/SDP packets (1 from asterisk to openser and 1
from openser to client)
AAA.AAA.AAA.AAA stands for IP of Asterisk
OOO.OOO.OOO.OOO stands for IP of OpenSER
CCC.CCC.CCC.CCC stands for IP of client
3333333333 is the called number
No. Time Source Destination Protocol Info
20 12.646925 AAA.AAA.AAA.AAA OOO.OOO.OOO.OOO SIP/SDP
Status: 200 OK, with session description
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP
OOO.OOO.OOO.OOO;branch=z9hG4bK5bbd.eaf4f093.0;received=OOO.OOO.OOO.OOO
Via: SIP/2.0/UDP
CCC.CCC.CCC.CCC:8952;branch=z9hG4bK-d87543-e15656230434101e-1--d87543-;rport=8952
Record-Route: <sip:OOO.OOO.OOO.OOO;lr=on;ftag=9043ec70>
From: "test"<sip:[EMAIL PROTECTED]>;tag=9043ec70
To: "3333333333"<sip:[EMAIL PROTECTED]>;tag=as30a7528b
Call-ID: 98684a222a2eeb7aYmVlZTUzZDRhNjMzN2Y0MTZhYmNmOTc5MzQ4OGI3ZGU.
CSeq: 3 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Type: application/sdp
Content-Length: 291
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 20137 20138 IN IP4
AAA.AAA.AAA.AAA
Session Name (s): session
Connection Information (c): IN IP4 AAA.AAA.AAA.AAA
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 58508
RTP/AVP 98 3 8 0 101
Media Attribute (a): rtpmap:98 iLBC/8000
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): silenceSupp:off - - - -
No. Time Source Destination Protocol Info
21 12.647437 OOO.OOO.OOO.OOO CCC.CCC.CCC.CCC SIP/SDP
Status: 200 OK, with session description
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP
OOO.OOO.OOO.OOO:5060;branch=z9hG4bK-d87543-e15656230434101e-1--d87543-;rport=8952
Record-Route: <sip:OOO.OOO.OOO.OOO;lr=on;ftag=9043ec70>
From: "test"<sip:[EMAIL PROTECTED]>;tag=9043ec70
To: "3333333333"<sip:[EMAIL PROTECTED]>;tag=as30a7528b
Call-ID: 98684a222a2eeb7aYmVlZTUzZDRhNjMzN2Y0MTZhYmNmOTc5MzQ4OGI3ZGU.
CSeq: 3 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Type: application/sdp
Content-Length: 291
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 20137 20138 IN IP4
OOO.OOO.OOO.OOO
Session Name (s): session
Connection Information (c): IN IP4 OOO.OOO.OOO.OOO
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 58508
RTP/AVP 98 3 8 0 101
Media Attribute (a): rtpmap:98 iLBC/8000
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): silenceSupp:off - - - -
Tnx very much for help again
Regards
Edoardo
regards
klaus
regards
klaus
Edoardo Serra wrote:
Hi guys,
I'm having a problem with an OpenSER acting as registrar
server and load balancer for many Asterisk servers.
In a few words: "users are registering on openser and, when they
want to make a call, OpenSER proxies the request to an Asterisk
server with the dispatcher module"
Here is the intended data flow (SIP goes through OpenSER and media
goes directly to Asterisk)
User <-- SIP --> OpenSER <-- SIP --> Asterisk
User <-- RTP --> Asterisk
Both, OpenSER and Asterisks have public IPs
I already have a working setup of that and everything seems working
correctly.
I'm trying to replicate that setup on another site, same
configurations of the boxes, same versions of OpenSER and Asterisk,
etc... but I'm having monodirectional Audio.
Having a look with tethereal I see that OpenSER, when the
communication is answered, sends a SIP packet (200 OK) to the user
indicating itself as media endpoint instead of the Asterisks.
From that moment I see RTP packets flowing from the client to OpenSER
This seems really strange to me because I just copied the same
configurations file from a working setup to the new installation.
Tnx in advance for help.
Regards
P.S.: Here is my openser.cfg
## $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
## simple quick-start config script
#
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
#log_stderror=no # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#children=4
#port=5060
fifo="/tmp/ser_fifo"
#uid=nobody
#gid=nobody
# ------------------ module loading ----------------------------------
loadmodule "/usr/lib/openser/modules/sl.so"
loadmodule "/usr/lib/openser/modules/tm.so"
loadmodule "/usr/lib/openser/modules/rr.so"
loadmodule "/usr/lib/openser/modules/maxfwd.so"
loadmodule "/usr/lib/openser/modules/usrloc.so"
loadmodule "/usr/lib/openser/modules/registrar.so"
loadmodule "/usr/lib/openser/modules/nathelper.so"
loadmodule "/usr/lib/openser/modules/textops.so"
loadmodule "/usr/lib/openser/modules/exec.so"
loadmodule "/usr/lib/openser/modules/uri.so"
loadmodule "/usr/lib/openser/modules/uri_db.so"
loadmodule "/usr/lib/openser/modules/dispatcher.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/openser/modules/mysql.so"
loadmodule "/usr/lib/openser/modules/auth.so"
loadmodule "/usr/lib/openser/modules/auth_db.so"
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url", "mysql://xxx:[EMAIL PROTECTED]/openser")
modparam("usrloc", "timer_interval", 120)
modparam("auth_db", "calculate_ha1", 0)
modparam("auth_db", "db_url", "mysql://xxx:[EMAIL PROTECTED]/voip")
modparam("uri_db", "db_url", "mysql://xxx:[EMAIL PROTECTED]/openser")
modparam("rr", "enable_full_lr", 1)
modparam("registrar", "nat_flag", 6)
modparam("registrar", "max_expires", 3600)
modparam("registrar", "min_expires", 60)
modparam("registrar", "append_branches", 0)
modparam("registrar", "desc_time_order", 1)
modparam("nathelper", "natping_interval", 20) # Ping interval 20 s
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
modparam("dispatcher", "force_dst", 1)
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
exit;
};
if ( (method=="OPTIONS") || (method=="SUBSCRIBE") || (method=="NOTIFY") ) {
sl_send_reply("405", "Method Not Allowed");
exit;
}
if (!method=="REGISTER") {
record_route();
};
if ((src_ip==xxx.xxx.xxx.xxx) || (src_ip==xxx.xxx.xxx.xxx)) { # IP
of Asterisks
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
exit;
};
if (nat_uac_test("3")) {
if ((method=="REGISTER") || (method=="INVITE") || (method=="OPTIONS")) {
fix_nated_contact();
force_rport();
setflag(6); # Mark as NATed
}
}
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (method=="REGISTER") {
if (!proxy_authorize("domain", "openser_view")) {
proxy_challenge("domain", "0");
exit;
}
if (!check_to()) {
sl_send_reply("403", "Digest username and URI username do NOT
match! Stay away!");
exit;
}
save("location");
exit;
};
if (method=="INVITE") {
if (!proxy_authorize("domain", "openser_view")) {
proxy_challenge("domain", "0");
exit;
}
if (!check_from()) {
sl_send_reply("403", "Digest username and URI username do NOT
match! Stay away!");
exit;
}
}
# loose-route processing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
exit;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
exit;
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# ! Nathelper
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
!search("^Route:")){
sl_send_reply("479", "We don't forward to private IP addresses");
exit;
};
# NAT processing of replies; apply to all transactions (for example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing); look at replies
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if ((src_ip!=xxx.xxx.xxx.xxx) && (src_ip!=xxx.xxx.xxx.xxx)) { # IP
of Asterisks
ds_select_dst("2", "0");
}
if (!t_relay()) {
sl_reply_error();
};
}
# ! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
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--
Klaus Darilion
nic.at
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