5 Apr
2005
5 Apr
'05
1:23 p.m.
Hello,
I have a problem whereby when a private client rings a public client only the public user
can hear voice and when the public user rings the private user, no audio is transmitted.
After looking at the messages I have have determined that the contact header and sdp part
of the invite contains the private address of the natted client. This would lead me to
believe that the registration process is incorrect. My problem is that I believe my script
should handle the registration process correctly and I suspect that the following code is
being skipped: I tried changing it to nat_uac_test("19") and
fix_nated_sdp("3") but that didnt help.
if (nat_uac_test("3")){
if (method == "REGISTER" || ! search("^Record-Route:")){
log("Log: Someone trying to register from private IP,rewriting\n");
fix_nated_contact(); #Rewrite contact with source IP
if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
I have included the log message below and my ser.cfg as an attachment. Please let me know
where I could be going wrong.
Thank you,
Vivienne.
2093 (private) ringing 2092 (public)
U 84.203.148.14:5060 -> 84.203.148.146:5060
INVITE sip:2092@84.203.148.146 SIP/2.0..Via: SIP/2.0/UDP 172.16.3.31;branch
=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc376
dcd4655094..To: <sip:2092@84.203.148.146>..Contact: <sip:2093@172.16.3.31>.
.Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327 IN
VITE..User-Agent: Grandstream BT100 1.0.5.18..Max-Forwards: 70..Allow: INVI
TE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: applic
ation/sdp..Content-Length: 362....v=0..o=2093 8000 0 IN IP4 172.16.3.31..s=
SIP Call..c=IN IP4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP 0 8 4 18 2 15 9
9 9..a=sendrecv..a=rtpmap:0 PCMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G729/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:
15 G728/8000/3..a=rtpmap:99 iLBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9 G722
/8000/3..a=ptime:20..
U 84.203.148.146:5060 -> 84.203.148.14:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 172.16
.3.31;branch=z9hG4bK1a48edc121f5bc1f;rport=5060;received=84.203.148.14..Fro
m: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.
203.148.146>..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..Se
rver: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning:
392 84.203.148.146:5060 "Noisy feedback tells: pid=4732 req_src_ip=84.203
.148.14 req_src_port=5060 in_uri=sip:2092@84.203.148.146 out_uri=sip:2092@1
57.190.74.151 via_cnt==1"....
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP 84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.0..Via: SIP/2.0/UDP 172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To: <sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID: 44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100 1.0.5.18..
Max-Forwards: 69..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length: 443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4 84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99 9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4 G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15 G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9 G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 84.203.148.146:5060 -> 157.190.74.150:5060
INVITE sip:2092@157.190.74.150 SIP/2.0..Via: SIP/2.0/UDP 84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.1..Via: SIP/2.0/UDP 172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To: <sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID: 44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100 1.0.5.18..
Max-Forwards: 69..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length: 443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4 84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99 9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4 G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15 G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9 G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP 84.203.148.146;branch=z9hG4bK77bc.b54c
a216.0..Via: SIP/2.0/UDP 172.16.3.31;rport=5060;received=84.203.148.14;bran
ch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc3
76dcd4655094..To: <sip:2092@84.203.148.146>..Call-ID: 44e1ae63c476fbff@172.
16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100 1.0.5.18..Conten
t-Length: 0....
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 84.203.148.146;branch=z9hG4bK77bc.b54
ca216.0..Via: SIP/2.0/UDP 172.16.3.31;rport=5060;received=84.203.148.14;bra
nch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc
376dcd4655094..To: <sip:2092@84.203.148.146>;tag=10bdf2044401a257..Call-ID:
44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream
BT100 1.0.5.18..Content-Length: 0....
Send instant messages to your online friends http://uk.messenger.yahoo.com
#
# $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)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
alias="157.190.74.152:5060"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
#loadmodule "/usr/lib/ser/modules/pa.so"
loadmodule "/usr/lib/ser/modules/cpl-c.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
#NB Had to up this value from 1 to 11 because reinvites were bombarding called phone
modparam("rr", "enable_full_lr", 11)
#!! Nathelper
modparam("registrar", "nat_flag", 60)
modparam("nathelper", "natping_interval", 30) #Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) #Ping only clients
behind NAT
modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock")
modparam("tm", "fr_inv_timer", 20)
#modparam("pa", "default_expires", 3600)
modparam("usrloc", "db_url",
"sql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_db",
"mysql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_table", "cpl")
#modparam("cpl-c", "cpl_dtd_file",
"/work/sip_router/modules/cpl-ccpl-06.dtd")
# ------------------------- 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");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
#######################################################################################
if (nat_uac_test("3")){
if (method == "REGISTER" || ! search("^Record-Route:")){
log("Log: Someone trying to register from private IP,rewriting\n");
fix_nated_contact(); #Rewrite contact with source IP
if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
########################################################################################
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (method =="REGISTER") record_route();
# loose-route processing
if (loose_route()) {
#commented 11/02/05
#t_relay();
route(1);
break;
};
# 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 (uri==myself) {
log(1,"into loop");
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("157.190.74.152", "subscriber")) {
# www_challenge("157.190.74.152", "0");
# break;
# };
#cpl_process_register();
save("location");
break;
};
# if (method=="SUBSCRIBE")
# {
# log(1, "Subscribe\n");
# if(t_newtran())
# {
# log(1, "Registrar\n");
# handle_subscription("registrar");
# };
# break;
# };
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
if (method=="INVITE"){
log(1,"in invite loop");
#break; #no 100 trying
#if (!cpl_run_script("incoming","is_stateless"))
#{
# #script execution failed
# t_reply("500", "CPL script execution failed");
#};
t_on_failure("1");
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
#sl_send_reply("404", "Not Found");
route(2);
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
#commented 11/02/05#######################
if (!t_relay()) {
sl_reply_error();
};
#route(1);
}
######################################entered
11/02/05############################################################
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");
break;
};
if (isflagset(6)){
force_rtp_proxy();
}
t_on_reply("1");
if(!t_relay()){
sl_reply_error();
break;
};
}
######################################entered
11/02/05############################################################
#!! Nathelper
onreply_route[1] {
if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){
fix_nated_contact();
force_rtp_proxy();
}else if (nat_uac_test("1")){
fix_nated_contact();
};
}
#################################################################################################################
# ------------- handling of unavailable user ------------------
route[2] {
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method ==
"CANCEL")) {
sl_send_reply("404", "Not Found");
break;
};
# forward to voicemail now
rewritehostport("157.190.74.152:5062");
t_relay_to_udp("157.190.74.152", "5062");
}
# if forwarding downstream did not succeed, try voicemail running
# at 172.16.2.120:5062
failure_route[1] {
revert_uri();
rewritehostport("157.190.74.152:5062");
append_branch();
t_relay_to_udp("157.190.74.152", "5062");
}
5 Apr
5 Apr
1:53 p.m.
Vivienne,
This is the first INVITE going from SER to your public phone. I have prefixed my comments
with *==>
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54ca216.0..
Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..
From: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..
To: <sip:2092@84.203.148.146>..
Contact: <sip:2093@84.203.148.14:5060>..
*==> Correctly changed to the public address and port of 2093
Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..
User-Agent: Grandstream BT100 1.0.5.18..
Max-Forwards: 69..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..
Content-Type: application/sdp..
Content-Length: 443....v=0..
o=2093 8000 0 IN IP4 172.16.3.31..
s=SIP Call..
c=IN IP4 84.203.148.1484.203.148.146..
*==> Here it seems that first fix_nated_sdp("3") is called (replace private
IP with public IP), then force_rtp_proxy() (replace with proxy IP). You must only call
one of them (fix_nated_sdp("1") is ok)
t=0 0..
m=audio 35006 RTP/AVP 0 8 4 18 2 15 99 9..
a=sendrecv..
a=rtpmap:0 PCMU/8000/3..
a=rtpmap:8 PCMA/8000/3..
a=rtpmap:4 G723/8000/3..a=rtpmap:18 G729/8000/3..
a=rtpmap:2 G726-32/8000/3..
a=rtpmap:15 G728/8000/3..
a=rtpmap:99 iLBC/8000/3..
a=fmtp:99 mode=20..
a=rtpmap:9 G722/8000/3..
a=ptime:20..
a=direction:active..
*==> Added by fix_nated_sdp("1") (and "3")
a=oldmediaip:172.16.3.31..
*==> Added by fix_nated_sdp("3")
a=nortpproxy:yes..
*==> Added by force_rtp_proxy()
So, your call is proxied. Try using fix_nated_sdp("1"). It should make the
INVITE correct. You have not posted the OK, so I don't know what is happening there.
If you have followed the rtpproxy ONsip.org example in your onreply_route, you should be
fine.
Good luck!
g-)
---- Original Message ----
From: Vivienne Curran
To: serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 01:23 PM
Subject: [Serusers] Contact Header and SDP not rewritten
Hello,
I have a problem whereby when a private client rings a public client
only the public user can hear voice and when the public user rings
the private user, no audio is transmitted. After looking at the
messages I have have determined that the contact header and sdp part
of the invite contains the private address of the natted client. This
would lead me to believe that the registration process is incorrect.
My problem is that I believe my script should handle the registration
process correctly and I suspect that the following code is being
skipped: I tried changing it to nat_uac_test("19") and
fix_nated_sdp("3") but that didnt help.
if (nat_uac_test("3")){
if (method == "REGISTER" || ! search("^Record-Route:")){
log("Log: Someone trying to register from private IP,rewriting\n");
fix_nated_contact(); #Rewrite contact with source IP
if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
I have included the log message below and my ser.cfg as an
attachment. Please let me know where I could be going wrong.
Thank you,
Vivienne.
2093 (private) ringing 2092 (public)
U 84.203.148.14:5060 -> 84.203.148.146:5060
INVITE sip:2092@84.203.148.146 SIP/2.0..Via: SIP/2.0/UDP
172.16.3.31;branch
=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc376
dcd4655094..To: <sip:2092@84.203.148.146>..Contact:
<sip:2093@172.16.3.31>.
.Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:
10327 IN
VITE..User-Agent: Grandstream BT100 1.0.5.18..Max-Forwards:
70..Allow: INVI
TE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:
applic
ation/sdp..Content-Length: 362....v=0..o=2093 8000 0 IN IP4
172.16.3.31..s=
SIP Call..c=IN IP4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP 0 8 4
18 2 15 9
9 9..a=sendrecv..a=rtpmap:0 PCMU/8000/3..a=rtpmap:8
PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G729/8000/3..a=rtpmap:2
G726-32/8000/3..a=rtpmap:
15 G728/8000/3..a=rtpmap:99 iLBC/8000/3..a=fmtp:99
mode=20..a=rtpmap:9 G722
/8000/3..a=ptime:20..
U 84.203.148.146:5060 -> 84.203.148.14:5060
SIP/2.0 100 trying -- your call is important to us..Via:
SIP/2.0/UDP 172.16
.3.31;branch=z9hG4bK1a48edc121f5bc1f;rport=5060;received=84.203.148.14..Fro
m: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.
203.148.146>..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327
INVITE..Se
rver: Sip EXpress router (0.8.14 (i386/linux))..Content-Length:
0..Warning:
392 84.203.148.146:5060 "Noisy feedback tells: pid=4732
req_src_ip=84.203
.148.14 req_src_port=5060 in_uri=sip:2092@84.203.148.146
out_uri=sip:2092@1
57.190.74.151 via_cnt==1"....
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 84.203.148.146:5060 -> 157.190.74.150:5060
INVITE sip:2092@157.190.74.150 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.1..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54c
a216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bran
ch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc3
76dcd4655094..To: <sip:2092@84.203.148.146>..Call-ID:
44e1ae63c476fbff@172.
16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..Conten
t-Length: 0....
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54
ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bra
nch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc
376dcd4655094..To:
<sip:2092@84.203.148.146>;tag=10bdf2044401a257..Call-ID:
44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..User-Agent:
Grandstream
BT100 1.0.5.18..Content-Length: 0....
Send instant messages to your online friends
http://uk.messenger.yahoo.com
#
# $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)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
alias="157.190.74.152:5060"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
#loadmodule "/usr/lib/ser/modules/pa.so"
loadmodule "/usr/lib/ser/modules/cpl-c.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
#NB Had to up this value from 1 to 11 because reinvites were
bombarding called phone
modparam("rr", "enable_full_lr", 11)
#!! Nathelper
modparam("registrar", "nat_flag", 60)
modparam("nathelper", "natping_interval", 30) #Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) #Ping only clients
behind NAT
modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock")
modparam("tm", "fr_inv_timer", 20)
#modparam("pa", "default_expires", 3600)
modparam("usrloc", "db_url",
"sql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_db",
"mysql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_table", "cpl")
#modparam("cpl-c", "cpl_dtd_file",
"/work/sip_router/modules/cpl-ccpl-06.dtd")
# ------------------------- 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");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
#######################################################################################
if (nat_uac_test("3")){
if (method == "REGISTER" || ! search("^Record-Route:")){
log("Log: Someone trying to register from private IP,rewriting\n");
fix_nated_contact(); #Rewrite contact with source IP
if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
########################################################################################
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (method =="REGISTER") record_route();
# loose-route processing
if (loose_route()) {
#commented 11/02/05
#t_relay();
route(1);
break;
};
# 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 (uri==myself) {
log(1,"into loop");
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("157.190.74.152", "subscriber")) {
# www_challenge("157.190.74.152", "0");
# break;
# };
#cpl_process_register();
save("location");
break;
};
# if (method=="SUBSCRIBE")
# {
# log(1, "Subscribe\n");
# if(t_newtran())
# {
# log(1, "Registrar\n");
# handle_subscription("registrar");
# };
# break;
# };
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
if (method=="INVITE"){
log(1,"in invite loop");
#break; #no 100 trying
#if (!cpl_run_script("incoming","is_stateless"))
#{
# #script execution failed
# t_reply("500", "CPL script execution failed");
#};
t_on_failure("1");
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
#sl_send_reply("404", "Not Found");
route(2);
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
#commented 11/02/05#######################
if (!t_relay()) {
sl_reply_error();
};
#route(1);
}
######################################entered
11/02/05############################################################
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");
break;
};
if (isflagset(6)){
force_rtp_proxy();
}
t_on_reply("1");
if(!t_relay()){
sl_reply_error();
break;
};
}
######################################entered
11/02/05############################################################
#!! Nathelper
onreply_route[1] {
if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){
fix_nated_contact();
force_rtp_proxy();
}else if (nat_uac_test("1")){
fix_nated_contact();
};
}
#################################################################################################################
# ------------- handling of unavailable user ------------------
route[2] {
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method ==
"CANCEL")) {
sl_send_reply("404", "Not Found");
break;
};
# forward to voicemail now
rewritehostport("157.190.74.152:5062");
t_relay_to_udp("157.190.74.152", "5062");
}
# if forwarding downstream did not succeed, try voicemail running
# at 172.16.2.120:5062
failure_route[1] {
revert_uri();
rewritehostport("157.190.74.152:5062");
append_branch();
t_relay_to_udp("157.190.74.152", "5062");
}
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
2:25 p.m.
Greger,
Since fix_nated_register does not exist with 0.8.14, will fix_nated_contact do instead?
Also if I am leaving out the has_totag() at the start of the script, will this greatly
effect its functionality?
Thank you,
Vivienne
"Greger V. Teigre" <greger(a)teigre.com> wrote:
Vivienne,
This is the first INVITE going from SER to your public phone. I have prefixed my comments
with *==>
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54ca216.0..
Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..
From: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..
To: <sip:2092@84.203.148.146>..
Contact: <sip:2093@84.203.148.14:5060>..
*==> Correctly changed to the public address and port of 2093
Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..
User-Agent: Grandstream BT100 1.0.5.18..
Max-Forwards: 69..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..
Content-Type: application/sdp..
Content-Length: 443....v=0..
o=2093 8000 0 IN IP4 172.16.3.31..
s=SIP Call..
c=IN IP4 84.203.148.1484.203.148.146..
*==> Here it seems that first fix_nated_sdp("3") is called (replace private
IP with public IP), then force_rtp_proxy() (replace with proxy IP). You must only call
one of them (fix_nated_sdp("1") is ok)
t=0 0..
m=audio 35006 RTP/AVP 0 8 4 18 2 15 99 9..
a=sendrecv..
a=rtpmap:0 PCMU/8000/3..
a=rtpmap:8 PCMA/8000/3..
a=rtpmap:4 G723/8000/3..a=rtpmap:18 G729/8000/3..
a=rtpmap:2 G726-32/8000/3..
a=rtpmap:15 G728/8000/3..
a=rtpmap:99 iLBC/8000/3..
a=fmtp:99 mode=20..
a=rtpmap:9 G722/8000/3..
a=ptime:20..
a=direction:active..
*==> Added by fix_nated_sdp("1") (and "3")
a=oldmediaip:172.16.3.31..
*==> Added by fix_nated_sdp("3")
a=nortpproxy:yes..
*==> Added by force_rtp_proxy()
So, your call is proxied. Try using fix_nated_sdp("1"). It should make the
INVITE correct. You have not posted the OK, so I don't know what is happening there.
If you have followed the rtpproxy ONsip.org example in your onreply_route, you should be
fine.
Good luck!
g-)
---- Original Message ----
From: Vivienne Curran
To: serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 01:23 PM
Subject: [Serusers] Contact Header and SDP not rewritten
Hello,
I have a problem whereby when a private client rings a public client
only the public user can hear voice and when the public user rings
the private user, no audio is transmitted. After looking at the
messages I have have determined that the contact header and sdp part
of the invite contains the private address of the natted client. This
would lead me to believe that the registration process is incorrect.
My problem is that I believe my script should handle the registration
process correctly and I suspect that the following code is being
skipped: I tried changing it to nat_uac_test("19") and
fix_nated_sdp("3") but that didnt help.
if (nat_uac_test("3")){
if (method == "REGISTER" || ! search("^Record-Route:")){
log("Log: Someone trying to register from private IP,rewriting\n");
fix_nated_contact(); #Rewrite contact with source IP
if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
I have included the log message below and my ser.cfg as an
attachment. Please let me know where I could be going wrong.
Thank you,
Vivienne.
2093 (private) ringing 2092 (public)
U 84.203.148.14:5060 -> 84.203.148.146:5060
INVITE sip:2092@84.203.148.146 SIP/2.0..Via: SIP/2.0/UDP
172.16.3.31;branch
=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc376
dcd4655094..To: <sip:2092@84.203.148.146>..Contact:
<sip:2093@172.16.3.31>.
.Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:
10327 IN
VITE..User-Agent: Grandstream BT100 1.0.5.18..Max-Forwards:
70..Allow: INVI
TE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:
applic
ation/sdp..Content-Length: 362....v=0..o=2093 8000 0 IN IP4
172.16.3.31..s=
SIP Call..c=IN IP4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP 0 8 4
18 2 15 9
9 9..a=sendrecv..a=rtpmap:0 PCMU/8000/3..a=rtpmap:8
PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G729/8000/3..a=rtpmap:2
G726-32/8000/3..a=rtpmap:
15 G728/8000/3..a=rtpmap:99 iLBC/8000/3..a=fmtp:99
mode=20..a=rtpmap:9 G722
/8000/3..a=ptime:20..
U 84.203.148.146:5060 -> 84.203.148.14:5060
SIP/2.0 100 trying -- your call is important to us..Via:
SIP/2.0/UDP 172.16
.3.31;branch=z9hG4bK1a48edc121f5bc1f;rport=5060;received=84.203.148.14..Fro
m: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.
203.148.146>..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327
INVITE..Se
rver: Sip EXpress router (0.8.14 (i386/linux))..Content-Length:
0..Warning:
392 84.203.148.146:5060 "Noisy feedback tells: pid=4732
req_src_ip=84.203
.148.14 req_src_port=5060 in_uri=sip:2092@84.203.148.146
out_uri=sip:2092@1
57.190.74.151 via_cnt==1"....
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 84.203.148.146:5060 -> 157.190.74.150:5060
INVITE sip:2092@157.190.74.150 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.1..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54c
a216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bran
ch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc3
76dcd4655094..To: <sip:2092@84.203.148.146>..Call-ID:
44e1ae63c476fbff@172.
16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..Conten
t-Length: 0....
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54
ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bra
nch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc
376dcd4655094..To:
<sip:2092@84.203.148.146>;tag=10bdf2044401a257..Call-ID:
44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..User-Agent:
Grandstream
BT100 1.0.5.18..Content-Length: 0....
Send instant messages to your online friends
http://uk.messenger.yahoo.com
#
# $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)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
alias="157.190.74.152:5060"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
#loadmodule "/usr/lib/ser/modules/pa.so"
loadmodule "/usr/lib/ser/modules/cpl-c.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
#NB Had to up this value from 1 to 11 because reinvites were
bombarding called phone
modparam("rr", "enable_full_lr", 11)
#!! Nathelper
modparam("registrar", "nat_flag", 60)
modparam("nathelper", "natping_interval", 30) #Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) #Ping only clients
behind NAT
modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock")
modparam("tm", "fr_inv_timer", 20)
#modparam("pa", "default_expires", 3600)
modparam("usrloc", "db_url",
"sql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_db",
"mysql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_table", "cpl")
#modparam("cpl-c", "cpl_dtd_file",
"/work/sip_router/modules/cpl-ccpl-06.dtd")
# ------------------------- 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");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
#######################################################################################
if (nat_uac_test("3")){
if (method == "REGISTER" || ! search("^Record-Route:")){
log("Log: Someone trying to register from private IP,rewriting\n");
fix_nated_contact(); #Rewrite contact with source IP
if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
########################################################################################
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (method =="REGISTER") record_route();
# loose-route processing
if (loose_route()) {
#commented 11/02/05
#t_relay();
route(1);
break;
};
# 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 (uri==myself) {
log(1,"into loop");
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("157.190.74.152", "subscriber")) {
# www_challenge("157.190.74.152", "0");
# break;
# };
#cpl_process_register();
save("location");
break;
};
# if (method=="SUBSCRIBE")
# {
# log(1, "Subscribe\n");
# if(t_newtran())
# {
# log(1, "Registrar\n");
# handle_subscription("registrar");
# };
# break;
# };
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
if (method=="INVITE"){
log(1,"in invite loop");
#break; #no 100 trying
#if (!cpl_run_script("incoming","is_stateless"))
#{
# #script execution failed
# t_reply("500", "CPL script execution failed");
#};
t_on_failure("1");
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
#sl_send_reply("404", "Not Found");
route(2);
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
#commented 11/02/05#######################
if (!t_relay()) {
sl_reply_error();
};
#route(1);
}
######################################entered
11/02/05############################################################
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");
break;
};
if (isflagset(6)){
force_rtp_proxy();
}
t_on_reply("1");
if(!t_relay()){
sl_reply_error();
break;
};
}
######################################entered
11/02/05############################################################
#!! Nathelper
onreply_route[1] {
if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){
fix_nated_contact();
force_rtp_proxy();
}else if (nat_uac_test("1")){
fix_nated_contact();
};
}
#################################################################################################################
# ------------- handling of unavailable user ------------------
route[2] {
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method ==
"CANCEL")) {
sl_send_reply("404", "Not Found");
break;
};
# forward to voicemail now
rewritehostport("157.190.74.152:5062");
t_relay_to_udp("157.190.74.152", "5062");
}
# if forwarding downstream did not succeed, try voicemail running
# at 172.16.2.120:5062
failure_route[1] {
revert_uri();
rewritehostport("157.190.74.152:5062");
append_branch();
t_relay_to_udp("157.190.74.152", "5062");
}
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Send instant messages to your online friends http://uk.messenger.yahoo.com
2:38 p.m.
Yes, you can use fix_nated_contact instead. It is not entirely RFC-compliant, but
that's what you have in 0.8.14.
The has_totag() only tests to see if the INVITE has a To header, which means that it is
in-dialog and thus is a re-INVITE. An INVITE will normally not have loose routing unless
you have another SIP proxy forwarding an INVITE to you (in which case you should assume
that the other proxy handles NAT and thus not trigger NAT-related code). You can safely
remove the has_totag() if you use force_rtp_proxy("l")
g-)
---- Original Message ----
From: Vivienne Curran
To: Greger V. Teigre ; serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 02:25 PM
Subject: Re: [Serusers] Contact Header and SDP not rewritten
Greger,
Since fix_nated_register does not exist with 0.8.14, will
fix_nated_contact do instead? Also if I am leaving out the
has_totag() at the start of the script, will this greatly effect its
functionality?
Thank you,
Vivienne
"Greger V. Teigre" <greger(a)teigre.com> wrote:
Vivienne,
This is the first INVITE going from SER to your public phone. I have
prefixed my comments with *==>
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54ca216.0..
Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..
From: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..
To: <sip:2092@84.203.148.146>..
Contact: <sip:2093@84.203.148.14:5060>..
*==> Correctly changed to the public address and port of 2093
Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:
10327 INVITE..
User-Agent: Grandstream BT100 1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..
Content-Type: application/sdp..
Content-Length: 443....v=0..
o=2093 8000 0 IN IP4 172.16.3.31..
s=SIP Call..
c=IN IP4 84.203.148.1484.203.148.146..
*==> Here it seems that first fix_nated_sdp("3") is called (replace
private IP with public IP), then force_rtp_proxy() (replace with
proxy IP). You must only call one of them (fix_nated_sdp("1") is ok)
t=0 0..
m=audio 35006 RTP/AVP 0 8 4 18 2 15 99 9..
a=sendrecv..
a=rtpmap:0 PCMU/8000/3..
a=rtpmap:8 PCMA/8000/3..
a=rtpmap:4 G723/8000/3..a=rtpmap:18 G729/8000/3..
a=rtpmap:2 G726-32/8000/3..
a=rtpmap:15 G728/8000/3..
a=rtpmap:99 iLBC/8000/3..
a=fmtp:99 mode=20..
a=rtpmap:9 G722/8000/3..
a=ptime:20..
a=direction:active..
*==> Added by fix_nated_sdp("1") (and "3")
a=oldmediaip:172.16.3.31..
*==> Added by fix_nated_sdp("3")
a=nortpproxy:yes..
*==> Added by force_rtp_proxy()
So, your call is proxied. Try using fix_nated_sdp("1"). It should
make the INVITE correct. You have not posted the OK, so I don't know
what is happening there. If you have followed the rtpproxy ONsip.org
example in your onreply_route, you should be fine.
Good luck!
g-)
---- Original Message ----
From: Vivienne Curran
To: serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 01:23 PM
Subject: [Serusers] Contact Header and SDP not rewritten
Hello,
I have a problem whereby when a private client rings a public client
only the public user can hear voice and when the public user rings
the private user, no audio is transmitted. After looking at the
messages I have have determined that the contact header and sdp part
of the invite contains the private address of the natted client. This
would lead me to believe that the registration process is incorrect.
My problem is that I believe my script should handle the registration
process correctly and I suspect that the following code is being
skipped: I tried changing it to nat_uac_test("19") and
fix_nated_sdp("3") but that didnt help.
Send instant messages to your
online friends
http://uk.messenger.yahoo.com if (nat_uac_test("3")){
if
(method == "REGISTER" || ! search("^Record-Route:")){
log("Log: Someone trying to register from private
IP,rewriting\n"); fix_nated_contact(); #Rewrite contact with
source IP if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
I have included the log message below and my ser.cfg as an
attachment. Please let me know where I could be going wrong.
Thank you,
Vivienne.
2093 (private) ringing 2092 (public)
U 84.203.148.14:5060 -> 84.203.148.146:5060
! INVITE sip:2092@84.203.148.146 SIP/2.0..Via: SIP/2.0/UDP
172.16.3.31;branch
=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc376
dcd4655094..To: <sip:2092@84.203.148.146>..Contact:
<sip:2093@172.16.3.31>.
.Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:
10327 IN
VITE..User-Agent: Grandstream BT100 1.0.5.18..Max-Forwards:
70..Allow: INVI
TE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:
applic
ation/sdp..Content-Length: 362....v=0..o=2093 8000 0 IN IP4
172.16.3.31..s=
SIP Call..c=IN IP4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP 0 8 4
18 2 15 9
9 9..a=sendrecv..a=rtpmap:0 PCMU/8000/3..a=rtpmap:8
PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rt! pmap:18 G729/8000/3..a=rtpmap:2
G726-32/8000/3..a=rtpmap:
15 G728/8000/3..a=rtpmap:99 iLBC/8000/3..a=fmtp:99
mode=20..a=rtpmap:9 G722
/8000/3..a=ptime:20..
U 84.203.148.146:5060 -> 84.203.148.14:5060
SIP/2.0 100 trying -- your call is important to us..Via:
SIP/2.0/UDP 172.16
.3.31;branch=z9hG4bK1a48edc121f5bc1f;rport=5060;received=84.203.148.14..Fro
m: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.
203.148.146>..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327
INVITE..Se
rver: Sip EXpress router (0.8.14 (i386/linux))..Content-Length:
0..Warning:
392 84.203.148.146:5060 "Noisy feedback tells: pid=4732
req_src_ip=84.203
.148.14 req_src_port=5060 in_uri=sip:2092@84.203.148.146
out_uri=sip:2092@1
57.190.74.151 via_cnt==1"....
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 84.203.148.146:5060 -> 157.190.74.150:5060
INVITE sip:2092@157.190.74.150 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.1..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54c
a216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bran
ch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc3
76dcd4655094..To: <sip:2092@84.203.148.146>..Call-ID:
44e1ae63c476fbff@172.
16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..Conten
t-Length: 0....
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 180 Ringing! ..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54
ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bra
nch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc
376dcd4655094..To:
<sip:2092@84.203.148.146>;tag=10bdf2044401a257..Call-ID:
44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..User-Agent:
Grandstream
BT100 1.0.5.18..Content-Length: 0....
Send instant messages to your online friends
http://uk.messenger.yahoo.com
#
# $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 # deb! ug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
alias="157.190.74.152:5060"
# ------------------ module loading
----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
#loadmodule "/usr/lib/ser/modules/pa.so"
loadmodule "/usr/lib/ser/modules/cpl-c.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters
---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
#NB Had to up this value from 1 to 11 because reinvites were
bombarding called phone
modparam("rr", "enable_full_lr", 11)
#!! Nathelper
modparam("registrar", "nat_flag", 60)
modparam("nathelper", "natping_interval", 30) #Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) #Ping only clients
behind NAT
modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock")
modparam("tm", "fr_inv_timer", 20)
#modparam("pa", "default_expires", 3600)
modparam("usrloc", "db_url",
"sql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_db",
"mysql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_table", "cpl")
#modparam("cpl-c", "cpl_dtd_file",
"/work/sip_router/modules/cpl-ccpl-06.dtd")
# ------------------------- 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");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
#######################################################################################
if (nat_uac_test("3")){
if (method == "REGISTER" || ! search("^Record-Route:")){
log("Log: So! meone trying to register from private IP,rewriting\n");
fix_nated_contact(); #Rewrite contact with source IP
if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
########################################################################################
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (method =="REGISTER") record_route();
# loose-route processing
if (loose_route()) {
#commented 11/02/05
#t_relay();
route(1);
break;
};
# 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 (uri==myself) {
log(1,"into loop");
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("157.190.74.152", "subscriber")) {
# www_challenge("157.190.74.152", "0");
# break;
# };
#cpl_process_register();
save("location");
break;
};
# if (method=="SUBSCRIBE")
# {
# log(1, "Subscribe\n");
# if(t_newtran())
# {
# log(1, "Registrar\n");
# handle_subscription("registrar");
# };
# break;
# };
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
if (method=="INVITE"){
log(1,"in ! invite loop");
#break; #no 100 trying
#if (!cpl_run_script("incoming","is_stateless"))
#{
# #script execution failed
# t_reply("500", "CPL script execution failed");
#};
t_on_failure("1");
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
#sl_send_reply("404", "Not Found");
route(2);
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
#commented 11/02/05#######################
if (!t_relay()) {
sl_reply_error();
};
#route(1);
}
######################################entered
11/02/05############################################################
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");
break;
};
if (isflagset(6)){
force_rtp_proxy();
}
t_on_reply("1");
if(!t_relay()){
sl_reply_error();
break;
};
}
######################################entered
11/02/05############################################################
#!! Nathelper
onreply_route[1] {
if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){
fix_nated_contact();
force_rtp_proxy();
}else if (nat_uac_test("1")){
fix_nated_contact();
};
}
#################################################################################################################
# ------------- handling of unavai! lable user ------------------
route[2] {
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method ==
"CANCEL")) {
sl_send_reply("404", "Not Found");
break;
};
# forward to voicemail now
rewritehostport("157.190.74.152:5062");
t_relay_to_udp("157.190.74.152", "5062");
}
# if forwarding downstream did not succeed, try voicemail running
# at 172.16.2.120:5062
failure_route[1] {
revert_uri();
rewritehostport("157.190.74.152:5062");
append_branch();
t_relay_to_udp("157.190.74.152", "5062");
}
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
5:22 p.m.
Thank you for that Greger. I have altered my script so that it exactly mimics the one in
the onsip document besides the has_totag and fix_nated register. All is good when I ring
from a private phone to a public phone i.e. the audio is very clear and the following
messages are in /var/log.
ERROR: extract_body: message body has length zero
ERROR: force_rtp_proxy2: cant extract body from the message.
I assume this is because of the 200 OK to a register message where theres no sdp?? Is this
correct?
However when I try to phone from public into private I get:
ERROR: send_rtpp_command: cant read reply from a RTP Proxy.
I find this confusing because I know the rtpproxy is working.
BR
Vivienne.
"Greger V. Teigre" <greger(a)teigre.com> wrote:
Yes, you can use fix_nated_contact instead. It is not entirely RFC-compliant, but
that's what you have in 0.8.14.
The has_totag() only tests to see if the INVITE has a To header, which means that it is
in-dialog and thus is a re-INVITE. An INVITE will normally not have loose routing unless
you have another SIP proxy forwarding an INVITE to you (in which case you should assume
that the other proxy handles NAT and thus not trigger NAT-related code). You can safely
remove the has_totag() if you use force_rtp_proxy("l")
g-)
---- Original Message ----
From: Vivienne Curran
To: Greger V. Teigre ; serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 02:25 PM
Subject: Re: [Serusers] Contact Header and SDP not rewritten
Greger,
Since fix_nated_register does not exist with 0.8.14, will
fix_nated_contact do instead? Also if I am leaving out the
has_totag() at the start of the script, will this greatly effect its
functionality?
Thank you,
Vivienne
"Greger V. Teigre" <greger(a)teigre.com> wrote:
Vivienne,
This is the first INVITE going from SER to your public phone. I have
prefixed my comments with *==>
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54ca216.0..
Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..
From: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..
To: <sip:2092@84.203.148.146>..
Contact: <sip:2093@84.203.148.14:5060>..
*==> Correctly changed to the public address and port of 2093
Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:
10327 INVITE..
User-Agent: Grandstream BT100 1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..
Content-Type: application/sdp..
Content-Length: 443....v=0..
o=2093 8000 0 IN IP4 172.16.3.31..
s=SIP Call..
c=IN IP4 84.203.148.1484.203.148.146..
*==> Here it seems that first fix_nated_sdp("3") is called (replace
private IP with public IP), then force_rtp_proxy() (replace with
proxy IP). You must only call one of them (fix_nated_sdp("1") is ok)
t=0 0..
m=audio 35006 RTP/AVP 0 8 4 18 2 15 99 9..
a=sendrecv..
a=rtpmap:0 PCMU/8000/3..
a=rtpmap:8 PCMA/8000/3..
a=rtpmap:4 G723/8000/3..a=rtpmap:18 G729/8000/3..
a=rtpmap:2 G726-32/8000/3..
a=rtpmap:15 G728/8000/3..
a=rtpmap:99 iLBC/8000/3..
a=fmtp:99 mode=20..
a=rtpmap:9 G722/8000/3..
a=ptime:20..
a=direction:active..
*==> Added by fix_nated_sdp("1") (and "3")
a=oldmediaip:172.16.3.31..
*==> Added by fix_nated_sdp("3")
a=nortpproxy:yes..
*==> Added by force_rtp_proxy()
So, your call is proxied. Try using fix_nated_sdp("1"). It should
make the INVITE correct. You have not posted the OK, so I don't know
what is happening there. If you have followed the rtpproxy ONsip.org
example in your onreply_route, you should be fine.
Good luck!
g-)
---- Original Message ----
From: Vivienne Curran
To: serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 01:23 PM
Subject: [Serusers] Contact Header and SDP not rewritten
Send instant messages to your online friends http://uk.messenger.yahoo.com
Hello,
I have a problem whereby when a private client rings a public client
only the public user can hear voice and when the public user rings
the private user, no audio is transmitted. After looking at the
messages I have have determined that the contact header and sdp part
of the invite contains the private address of the natted client. This
would lead me to believe that the registration process is incorrect.
My problem is that I believe my script should handle the registration
process correctly and I suspect that the following code is being
skipped: I tried changing it to nat_uac_test("19") and
fix_nated_sdp("3") but that didnt help.
Send instant messages to your
online friends
http://uk.messenger.yahoo.com if (nat_uac_test("3")){
if
(method == "REGISTER" || ! search("^Record-Route:")){
log("Log: Someone trying to register from private
IP,rewriting\n"); fix_nated_contact(); #Rewrite contact with
source IP if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
I have included the log message below and my ser.cfg as an
attachment. Please let me know where I could be going wrong.
Thank you,
Vivienne.
2093 (private) ringing 2092 (public)
U 84.203.148.14:5060 -> 84.203.148.146:5060
! INVITE sip:2092@84.203.148.146 SIP/2.0..Via: SIP/2.0/UDP
172.16.3.31;branch
=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc376
dcd4655094..To: <sip:2092@84.203.148.146>..Contact:
<sip:2093@172.16.3.31>.
.Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:
10327 IN
VITE..User-Agent: Grandstream BT100 1.0.5.18..Max-Forwards:
70..Allow: INVI
TE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:
applic
ation/sdp..Content-Length: 362....v=0..o=2093 8000 0 IN IP4
172.16.3.31..s=
SIP Call..c=IN IP4 172.16.3.31..t=0 0..m=audio 5004 RTP/AVP 0 8 4
18 2 15 9
9 9..a=sendrecv..a=rtpmap:0 PCMU/8000/3..a=rtpmap:8
PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rt! pmap:18 G729/8000/3..a=rtpmap:2
G726-32/8000/3..a=rtpmap:
15 G728/8000/3..a=rtpmap:99 iLBC/8000/3..a=fmtp:99
mode=20..a=rtpmap:9 G722
/8000/3..a=ptime:20..
U 84.203.148.146:5060 -> 84.203.148.14:5060
SIP/2.0 100 trying -- your call is important to us..Via:
SIP/2.0/UDP 172.16
.3.31;branch=z9hG4bK1a48edc121f5bc1f;rport=5060;received=84.203.148.14..Fro
m: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.
203.148.146>..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327
INVITE..Se
rver: Sip EXpress router (0.8.14 (i386/linux))..Content-Length:
0..Warning:
392 84.203.148.146:5060 "Noisy feedback tells: pid=4732
req_src_ip=84.203
.148.14 req_src_port=5060 in_uri=sip:2092@84.203.148.146
out_uri=sip:2092@1
57.190.74.151 via_cnt==1"....
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 84.203.148.146:5060 -> 157.190.74.150:5060
INVITE sip:2092@157.190.74.150 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.1..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54c
a216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bran
ch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc3
76dcd4655094..To: <sip:2092@84.203.148.146>..Call-ID:
44e1ae63c476fbff@172.
16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..Conten
t-Length: 0....
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 180 Ringing! ..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54
ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bra
nch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc
376dcd4655094..To:
<sip:2092@84.203.148.146>;tag=10bdf2044401a257..Call-ID:
44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..User-Agent:
Grandstream
BT100 1.0.5.18..Content-Length: 0....
Send instant messages to your online friends
http://uk.messenger.yahoo.com
#
# $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 # deb! ug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
alias="157.190.74.152:5060"
# ------------------ module loading
----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
#loadmodule "/usr/lib/ser/modules/pa.so"
loadmodule "/usr/lib/ser/modules/cpl-c.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters
---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
#NB Had to up this value from 1 to 11 because reinvites were
bombarding called phone
modparam("rr", "enable_full_lr", 11)
#!! Nathelper
modparam("registrar", "nat_flag", 60)
modparam("nathelper", "natping_interval", 30) #Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) #Ping only clients
behind NAT
modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock")
modparam("tm", "fr_inv_timer", 20)
#modparam("pa", "default_expires", 3600)
modparam("usrloc", "db_url",
"sql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_db",
"mysql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_table", "cpl")
#modparam("cpl-c", "cpl_dtd_file",
"/work/sip_router/modules/cpl-ccpl-06.dtd")
# ------------------------- 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");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
#######################################################################################
if (nat_uac_test("3")){
if (method == "REGISTER" || ! search("^Record-Route:")){
log("Log: So! meone trying to register from private IP,rewriting\n");
fix_nated_contact(); #Rewrite contact with source IP
if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
########################################################################################
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (method =="REGISTER") record_route();
# loose-route processing
if (loose_route()) {
#commented 11/02/05
#t_relay();
route(1);
break;
};
# 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 (uri==myself) {
log(1,"into loop");
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("157.190.74.152", "subscriber")) {
# www_challenge("157.190.74.152", "0");
# break;
# };
#cpl_process_register();
save("location");
break;
};
# if (method=="SUBSCRIBE")
# {
# log(1, "Subscribe\n");
# if(t_newtran())
# {
# log(1, "Registrar\n");
# handle_subscription("registrar");
# };
# break;
# };
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
if (method=="INVITE"){
log(1,"in ! invite loop");
#break; #no 100 trying
#if (!cpl_run_script("incoming","is_stateless"))
#{
# #script execution failed
# t_reply("500", "CPL script execution failed");
#};
t_on_failure("1");
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
#sl_send_reply("404", "Not Found");
route(2);
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
#commented 11/02/05#######################
if (!t_relay()) {
sl_reply_error();
};
#route(1);
}
######################################entered
11/02/05############################################################
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");
break;
};
if (isflagset(6)){
force_rtp_proxy();
}
t_on_reply("1");
if(!t_relay()){
sl_reply_error();
break;
};
}
######################################entered
11/02/05############################################################
#!! Nathelper
onreply_route[1] {
if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){
fix_nated_contact();
force_rtp_proxy();
}else if (nat_uac_test("1")){
fix_nated_contact();
};
}
#################################################################################################################
# ------------- handling of unavai! lable user ------------------
route[2] {
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method ==
"CANCEL")) {
sl_send_reply("404", "Not Found");
break;
};
# forward to voicemail now
rewritehostport("157.190.74.152:5062");
t_relay_to_udp("157.190.74.152", "5062");
}
# if forwarding downstream did not succeed, try voicemail running
# at 172.16.2.120:5062
failure_route[1] {
revert_uri();
rewritehostport("157.190.74.152:5062");
append_branch();
t_relay_to_udp("157.190.74.152", "5062");
}
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
6 Apr
6 Apr
7:46 a.m.
See inline.
Thank you for that Greger. I have altered my script so
that it
exactly mimics the one in the onsip document besides the has_totag
and fix_nated register. All is good when I ring from a private phone
to a public phone i.e. the audio is very clear and the following
messages are in /var/log.
ERROR: extract_body: message body has length zero
ERROR: force_rtp_proxy2: cant extract body from the message.
I assume this is because of the 200 OK to a register message where
theres no sdp?? Is this correct?
That's correct. You will find code in the example configs where we test for an empty
body before calling force_rtp_proxy.
However when I try to phone from public into private I
get:
ERROR: send_rtpp_command: cant read reply from a RTP Proxy.
I find this confusing because I know the rtpproxy is working.
This means that rtpproxy is not responding to a particular message. I have heard some
people have had problems with the socket based communication. I only use UDP. This is what
you do to set up udp (22222 is default port):
modparam("nathelper", "rtpproxy_sock",
"udp:localhost:22222")
rtpproxy must be started with -s udp:*
g-)
BR
Vivienne.
"Greger V. Teigre" <greger(a)teigre.com> wrote:
Yes, you can use fix_nated_contact instead. It is not entirely
RFC-compliant, but that's what you have in 0.8.14.
The has_totag() only tests to see if the INVITE has a To header,
which means that it is in-dialog and thus is a re-INVITE. An INVITE
will normally not have loose routing unless you have another SIP
proxy forwarding an INVITE to you (in which case you should assume
that the other proxy handles NAT and thus not trigger NAT-related
code). You can safely remove the has_totag() if you use
force_rtp_proxy("l")
g-)
---- Original Message ----
From: Vivienne Curran
To: Greger V. Teigre ; serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 02:25 PM
Subject: Re: [Serusers] Contact Header and SDP not rewritten
Greger,
Since fix_nated_register does not exist with 0.8.14, will
fix_nated_contact do instead? Also if I am leaving out the
has_totag() at the start of the script, will this greatly effect its
functionality?
Thank you,
Vivienne
"Greger V. Teigre" <greger(a)teigre.com> wrote:
Vivienne,
This is the first INVITE going from SER to your public phone. I have
prefixed my comments with *==>
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54ca216.0..
Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..
From: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..
To: <sip:2092@84.203.148.146>..
Contact: <sip:2093@84.203.148.14:5060>..
*==> Correctly changed to the public address and port of 2093
Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:
10327 INVITE..
User-Agent: Grandstream BT100 1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..
Content-Type: application/sdp..
Content-Length: 443....v=0..
o=2093 8000 0 IN IP4 172.16.3.31..
s=SIP Call..
c=IN IP4 84.203.148.1484.203.148.146..
*==> Here it seems that first fix_nated_sdp("3") is called (replace
private IP with public IP), then force_rtp_proxy() (replace with
proxy IP). You must only call one of them (fix_nated_sdp! ("1") is
ok) t=0 0..
m=audio 35006 RTP/AVP 0 8 4 18 2 15 99 9..
a=sendrecv..
a=rtpmap:0 PCMU/8000/3..
a=rtpmap:8 PCMA/8000/3..
a=rtpmap:4 G723/8000/3..a=rtpmap:18 G729/8000/3..
a=rtpmap:2 G726-32/8000/3..
a=rtpmap:15 G728/8000/3..
a=rtpmap:99 iLBC/8000/3..
a=fmtp:99 mode=20..
a=rtpmap:9 G722/8000/3..
a=ptime:20..
a=direction:active..
*==> Added by fix_nated_sdp("1") (and "3")
a=oldmediaip:172.16.3.31..
*==> Added by fix_nated_sdp("3")
a=nortpproxy:yes..
*==> Added by force_rtp_proxy()
So, your call is proxied. Try using fix_nated_sdp("1"). It should
make the INVITE correct. You have not posted the OK, so I don't know
what is happening there. If you have followed the rtpproxy ONsip.org
example in your onreply_route, you should be fine.
Good luck!
&g! t; g-)
---- Original Message ----
From: Vivienne Curran
To: serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 01:23 PM
Subject: [Serusers] Contact Header and SDP not rewritten
Send instant messages to your online friends
http://uk.messenger.yahoo.com Hello,
I have a problem whereby when a private client rings a public client
only the public user can hear voice and when the public user rings
the private user, no audio is transmitted. After looking at the
messages I have have determined that the contact header and sdp part
of the invite contains the private address of the natted client.
This would lead me to believe that the registration process is
incorrect. My problem is that I believe my script should handle the
registration process correctly and I suspect that the following
code is being skipped: I tried changing it to nat_uac_test("19") and
fix_nated_sdp("3") but that didnt help.
&g! t;
if (nat_uac_test("3")){
if
(method == "REGISTER" || ! search("^Record-Route:")){
log("Log: Someone trying to register from private
IP,rewriting\n"); fix_nated_contact(); #Rewrite contact with
source IP if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
I have included the log message below and my ser.cfg as an
attachment. Please let me know where I could be going wrong.
Thank you,
Vivienne.
> 2093 (private) ringing 2092 (public)
U 84.203.148.14:5060 -> 84.203.148.146:5060
! INVITE sip:2092@84.203.148.146 SIP/2.0..Via: SIP/2.0/UDP
172.16.3.31;branch
=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc376
dcd4655094..To: <sip:2092@84.203.148.146>..Contact:
<sip:2093@172.16.3.31>.
.Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:
10327 IN
VITE..User-Agent: Grandstream BT100 1.0.5.18..Max-Forwards:
70..Allow: INVI
TE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:
applic
ation/sdp..Content-Length: 362....v=0..o=2093 8000 0 IN IP4
172.16.3.31..s=
SIP Call..c=IN IP4 172.16.3.31..t=0 0..m=audio 500! 4 RTP/AVP 0 8
4 18 2 15 9
9 9..a=sendrecv..a=rtpmap:0 PCMU/8000/3..a=rtpmap:8
PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rt! pmap:18 G729/8000/3..a=rtpmap:2
G726-32/8000/3..a=rtpmap:
15 G728/8000/3..a=rtpmap:99 iLBC/8000/3..a=fmtp:99
mode=20..a=rtpmap:9 G722
/8000/3..a=ptime:20..
U 84.203.148.146:5060 -> 84.203.148.14:5060
SIP/2.0 100 trying -- your call is important to us..Via:
SIP/2.0/UDP 172.16
.3.31;branch=z9hG4bK1a48edc121f5bc1f;rport=5060;received=84.203.148.14..Fro
m: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.
203.148.146>..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327
INVITE..Se
rver: Sip EXpress router (! 0.8.14 (i386/linux))..Content-Length:
0..Warning:
392 84.203.148.146:5060 "Noisy feedback tells: pid=4732
req_src_ip=84.203
.148.14 req_src_port=5060 in_uri=sip:2092@84.203.148.146
out_uri=sip:2092@1
57.190.74.151 via_cnt==1"....
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 84.203.148.146:5060 -> 157.190.74.150:5060
INVITE sip:2092@157.190.74.150 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.1..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54c
a216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bran
ch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc3
76dcd4655094..To: <sip:2092@84.203.148.146>..Call-ID:
44e1ae63c476fbff@172.
16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..Conten
t-Length: 0....
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 180 Ringing! ..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54
ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bra
nch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc
376dcd4655094..To:
<sip:2092@84.203.148.146>;tag=10bdf2044401a257..Call-ID:
44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..User-Agent:
Grandstream
BT100 1.0.5.18..Content-Length: 0....
Send instant messages to your online friends
http://uk.messenger.yahoo.com
#
# $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 # deb! ug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
alias="157.190.74.152:5060"
# ------------------ module loading
----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
#loadmodule "/usr/lib/ser/modules/pa.so"
loadmodule "/usr/lib/ser/modules/cpl-c.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters
---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
#NB Had to up this value from 1 to 11 because reinvites were
bombarding called phone
modparam("rr", "enable_full_lr", 11)
#!! Nathelper
modparam("registrar", "nat_flag", 60)
modparam("nathelper", "natping_interval", 30) #Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) #Ping only clients
behind NAT
modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock")
modparam("tm", "fr_inv_timer", 20)
#modparam("pa", "default_expires", 3600)
modparam("usrloc", "db_url",
"sql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_db",
"mysql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_table", "cpl")
#modparam("cpl-c", "cpl_dtd_file",
"/work/sip_router/modules/cpl-ccpl-06.dtd")
# ------------------------- 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");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
#######################################################################################
if (nat_uac_test("3")){
if (method == "REGISTER" || ! search("^Record-Route:")){
log("Log: So! meone trying to register from private
IP,rewriting\n"); fix_nated_contact(); #Rewrite contact with source
IP
if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
########################################################################################
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol if (method =="REGISTER")
record_route();
# loose-route processing
if (loose_route()) {
#commented 11/02/05
#t_relay();
route(1);
break;
};
# 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 (uri==myself) {
log(1,"into loop");
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("157.190.74.152", "subscriber")) {
# www_challenge("157.190.74.152", "0");
# break;
# };
#cpl_process_register();
save("location");
break;
};
# if (method=="SUBSCRIBE")
&g! t; # {
# log(1, "Subscribe\n");
# if(t_newtran())
# {
# log(1, "Registrar\n");
# handle_subscription("registrar");
# };
# break;
# };
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
if (method=="INVITE"){
log(1,"in ! invite loop");
#break; #no 100 trying
#if (!cpl_run_script("incoming","is_stateless"))
#{
# #script execution failed
# t_reply("500", "CPL script execution failed");
#};
t_on_failure("1");
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
#sl_send_reply("404", "Not Found");
route(2);
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
#commented 11/02/05#######################
if (!t_relay()) {
sl_reply_error();
};
#route(1);
}
######################################entered
11/02/05############################################################
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");
break;
};
if (isflagset(6)){
force_rtp_proxy();
}
t_on_reply("1");
if(!t_relay()){
sl_reply_error();
break;
};
}
######################################entered
11/02/05############################################################
#!! Nathelper
onreply_route[1] {
if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){
fix_nated_contact();
force_rtp_proxy();
}else if (nat_uac_test("1")){
fix_nated_contact();
};
}
#################################################################################################################
# ------------- handling of unavai! lable user ------------------
route[2] {
# non-Voip -- just send "off-line"
if (!(method == "INVITE" ! || method == "ACK" || method ==
"CANCEL")) {
sl_send_reply("404", "Not Found");
break;
};
# forward to voicemail now
rewritehostport("157.190.74.152:5062");
t_relay_to_udp("157.190.74.152", "5062");
}
# if forwarding downstream did not succeed, try voicemail running
# at 172.16.2.120:5062
failure_route[1] {
revert_uri();
rewritehostport("157.190.74.152:5062");
append_branch();
t_relay_to_udp("157.190.74.152", "5062");
}
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Send instant messages to your
online friends
http://uk.messenger.yahoo.com
12:50 p.m.
New subject: [Serusers] RTPProxy error just for private to public
I changed the line modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock") to modparam("nathelper",
"rtpproxy_sock", "udp:localhost:22222") and started the rtpproxy as
./rtpproxy -s udp from the relevant directory and this resulted in a series of
"rtpp_command: no response from rtpproxy" and rtpproxy temporarily
disabled" errors. If I return to the original modparam and start it as ./rtpproxy
then it works but like I said when the private client rings the public client, I get
"ERROR: send_rtpp_command: cant read reply from a RTP Proxy".
Any further ideas? Has anyone on the mailing list experienced this? I am using the script
given in the onsip getting started doc for 0.9.0. but am using ser 0.8.14.
BR,
Vivienne
"Greger V. Teigre" <greger(a)teigre.com> wrote:
See inline.
Thank you for that Greger. I have altered my script so
that it
exactly mimics the one in the onsip document besides the has_totag
and fix_nated register. All is good when I ring from a private phone
to a public phone i.e. the audio is very clear and the following
messages are in /var/log.
ERROR: extract_body: message body has length zero
ERROR: force_rtp_proxy2: cant extract body from the message.
I assume this is because of the 200 OK to a register message where
theres no sdp?? Is this correct?
That's correct. You will find code in the example configs where we test for an empty
body before calling force_rtp_proxy.
However when I try to phone from public into private I
get:
ERROR: send_rtpp_command: cant read reply from a RTP Proxy.
I find this confusing because I know the rtpproxy is working.
This means that rtpproxy is not responding to a particular message. I have heard some
people have had problems with the socket based communication. I only use UDP. This is what
you do to set up udp (22222 is default port):
modparam("nathelper", "rtpproxy_sock",
"udp:localhost:22222")
rtpproxy must be started with -s udp:*
g-)
BR
Vivienne.
"Greger V. Teigre" <greger(a)teigre.com> wrote:
Yes, you can use fix_nated_contact instead. It is not entirely
RFC-compliant, but that's what you have in 0.8.14.
The has_totag() only tests to see if the INVITE has a To header,
which means that it is in-dialog and thus is a re-INVITE. An INVITE
will normally not have loose routing unless you have another SIP
proxy forwarding an INVITE to you (in which case you should assume
that the other proxy handles NAT and thus not trigger NAT-related
code). You can safely remove the has_totag() if you use
force_rtp_proxy("l")
g-)
---- Original Message ----
From: Vivienne Curran
To: Greger V. Teigre ; serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 02:25 PM
Subject: Re: [Serusers] Contact Header and SDP not rewritten
Send instant messages to your online friends http://uk.messenger.yahoo.com
Greger,
Since fix_nated_register does not exist with 0.8.14, will
fix_nated_contact do instead? Also if I am leaving out the
has_totag() at the start of the script, will this greatly effect its
functionality?
Thank you,
Vivienne
"Greger V. Teigre" <greger(a)teigre.com> wrote:
Vivienne,
This is the first INVITE going from SER to your public phone. I have
prefixed my comments with *==>
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54ca216.0..
Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..
From: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..
To: <sip:2092@84.203.148.146>..
Contact: <sip:2093@84.203.148.14:5060>..
*==> Correctly changed to the public address and port of 2093
Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:
10327 INVITE..
User-Agent: Grandstream BT100 1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..
Content-Type: application/sdp..
Content-Length: 443....v=0..
o=2093 8000 0 IN IP4 172.16.3.31..
s=SIP Call..
c=IN IP4 84.203.148.1484.203.148.146..
*==> Here it seems that first fix_nated_sdp("3") is called (replace
private IP with public IP), then force_rtp_proxy() (replace with
proxy IP). You must only call one of them (fix_nated_sdp! ("1") is
ok) t=0 0..
m=audio 35006 RTP/AVP 0 8 4 18 2 15 99 9..
a=sendrecv..
a=rtpmap:0 PCMU/8000/3..
a=rtpmap:8 PCMA/8000/3..
a=rtpmap:4 G723/8000/3..a=rtpmap:18 G729/8000/3..
a=rtpmap:2 G726-32/8000/3..
a=rtpmap:15 G728/8000/3..
a=rtpmap:99 iLBC/8000/3..
a=fmtp:99 mode=20..
a=rtpmap:9 G722/8000/3..
a=ptime:20..
a=direction:active..
*==> Added by fix_nated_sdp("1") (and "3")
a=oldmediaip:172.16.3.31..
*==> Added by fix_nated_sdp("3")
a=nortpproxy:yes..
*==> Added by force_rtp_proxy()
So, your call is proxied. Try using fix_nated_sdp("1"). It should
make the INVITE correct. You have not posted the OK, so I don't know
what is happening there. If you have followed the rtpproxy ONsip.org
example in your onreply_route, you should be fine.
Good luck!
&g! t; g-)
---- Original Message ----
From: Vivienne Curran
To: serusers(a)lists.iptel.org
Sent: Tuesday, April 05, 2005 01:23 PM
Subject: [Serusers] Contact Header and SDP not rewritten
Send instant messages to your online friends
http://uk.messenger.yahoo.com Hello,
I have a problem whereby when a private client rings a public client
only the public user can hear voice and when the public user rings
the private user, no audio is transmitted. After looking at the
messages I have have determined that the contact header and sdp part
of the invite contains the private address of the natted client.
This would lead me to believe that the registration process is
incorrect. My problem is that I believe my script should handle the
registration process correctly and I suspect that the following
code is being skipped: I tried changing it to nat_uac_test("19") and
fix_nated_sdp("3") but that didnt help.
&g! t;
if (nat_uac_test("3")){
if
(method == "REGISTER" || ! search("^Record-Route:")){
log("Log: Someone trying to register from private
IP,rewriting\n"); fix_nated_contact(); #Rewrite contact with
source IP if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
I have included the log message below and my ser.cfg as an
attachment. Please let me know where I could be going wrong.
Thank you,
Vivienne.
> 2093 (private) ringing 2092 (public)
U 84.203.148.14:5060 -> 84.203.148.146:5060
! INVITE sip:2092@84.203.148.146 SIP/2.0..Via: SIP/2.0/UDP
172.16.3.31;branch
=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc376
dcd4655094..To: <sip:2092@84.203.148.146>..Contact:
<sip:2093@172.16.3.31>.
.Supported: replaces..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq:
10327 IN
VITE..User-Agent: Grandstream BT100 1.0.5.18..Max-Forwards:
70..Allow: INVI
TE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:
applic
ation/sdp..Content-Length: 362....v=0..o=2093 8000 0 IN IP4
172.16.3.31..s=
SIP Call..c=IN IP4 172.16.3.31..t=0 0..m=audio 500! 4 RTP/AVP 0 8
4 18 2 15 9
9 9..a=sendrecv..a=rtpmap:0 PCMU/8000/3..a=rtpmap:8
PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rt! pmap:18 G729/8000/3..a=rtpmap:2
G726-32/8000/3..a=rtpmap:
15 G728/8000/3..a=rtpmap:99 iLBC/8000/3..a=fmtp:99
mode=20..a=rtpmap:9 G722
/8000/3..a=ptime:20..
U 84.203.148.146:5060 -> 84.203.148.14:5060
SIP/2.0 100 trying -- your call is important to us..Via:
SIP/2.0/UDP 172.16
.3.31;branch=z9hG4bK1a48edc121f5bc1f;rport=5060;received=84.203.148.14..Fro
m: "2093" <sip:2093@84.203.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.
203.148.146>..Call-ID: 44e1ae63c476fbff@172.16.3.31..CSeq: 10327
INVITE..Se
rver: Sip EXpress router (! 0.8.14 (i386/linux))..Content-Length:
0..Warning:
392 84.203.148.146:5060 "Noisy feedback tells: pid=4732
req_src_ip=84.203
.148.14 req_src_port=5060 in_uri=sip:2092@84.203.148.146
out_uri=sip:2092@1
57.190.74.151 via_cnt==1"....
U 84.203.148.146:5060 -> 157.190.74.151:5060
INVITE sip:2092@157.190.74.151 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 84.203.148.146:5060 -> 157.190.74.150:5060
INVITE sip:2092@157.190.74.150 SIP/2.0..Via: SIP/2.0/UDP
84.203.148.146;bra
nch=z9hG4bK77bc.b54ca216.1..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;receive
d=84.203.148.14;branch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.2
03.148.146>;tag=2dc376dcd4655094..To:
<sip:2092@84.203.148.146>..Contact: <
sip:2093@84.203.148.14:5060>..Supported: replaces..Call-ID:
44e1ae63c476fbf
f@172.16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..
Max-Forwards: 69..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
BSCRIBE..Content-Type: application/sdp..Content-Length:
443....v=0..o=2093
8000 0 IN IP4 172.16.3.31..s=SIP Call..c=IN IP4
84.203.148.1484.203.148.146
..t=0 0..m=audio 35006 RTP/AVP 0 8 4 18 2 15 99
9..a=sendrecv..a=rtpmap:0 P
CMU/8000/3..a=rtpmap:8 PCMA/8000/3..a=rtpmap:4
G723/8000/3..a=rtpmap:18 G72
9/8000/3..a=rtpmap:2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99 i
LBC/8000/3..a=fmtp:99 mode=20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=direct
ion:active..a=oldmediaip:172.16.3.31..a=nortpproxy:yes..
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54c
a216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bran
ch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc3
76dcd4655094..To: <sip:2092@84.203.148.146>..Call-ID:
44e1ae63c476fbff@172.
16.3.31..CSeq: 10327 INVITE..User-Agent: Grandstream BT100
1.0.5.18..Conten
t-Length: 0....
U 157.190.74.151:5060 -> 84.203.148.146:5060
SIP/2.0 180 Ringing! ..Via: SIP/2.0/UDP
84.203.148.146;branch=z9hG4bK77bc.b54
ca216.0..Via: SIP/2.0/UDP
172.16.3.31;rport=5060;received=84.203.148.14;bra
nch=z9hG4bK1a48edc121f5bc1f..From: "2093"
<sip:2093@84.203.148.146>;tag=2dc
376dcd4655094..To:
<sip:2092@84.203.148.146>;tag=10bdf2044401a257..Call-ID:
44e1ae63c476fbff@172.16.3.31..CSeq: 10327 INVITE..User-Agent:
Grandstream
BT100 1.0.5.18..Content-Length: 0....
Send instant messages to your online friends
http://uk.messenger.yahoo.com
#
# $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 # deb! ug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
alias="157.190.74.152:5060"
# ------------------ module loading
----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
#loadmodule "/usr/lib/ser/modules/pa.so"
loadmodule "/usr/lib/ser/modules/cpl-c.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/lib/ser/modules/auth.so"
loadmodule "/usr/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters
---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
#NB Had to up this value from 1 to 11 because reinvites were
bombarding called phone
modparam("rr", "enable_full_lr", 11)
#!! Nathelper
modparam("registrar", "nat_flag", 60)
modparam("nathelper", "natping_interval", 30) #Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) #Ping only clients
behind NAT
modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock")
modparam("tm", "fr_inv_timer", 20)
#modparam("pa", "default_expires", 3600)
modparam("usrloc", "db_url",
"sql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_db",
"mysql://root:1alfa156@localhost/ser")
#modparam("cpl-c", "cpl_table", "cpl")
#modparam("cpl-c", "cpl_dtd_file",
"/work/sip_router/modules/cpl-ccpl-06.dtd")
# ------------------------- 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");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
#######################################################################################
if (nat_uac_test("3")){
if (method == "REGISTER" || ! search("^Record-Route:")){
log("Log: So! meone trying to register from private
IP,rewriting\n"); fix_nated_contact(); #Rewrite contact with source
IP
if (method == "INVITE"){
fix_nated_sdp("1"); #Add direction=active to SDP
force_rtp_proxy();
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as Nated
};
};
########################################################################################
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol if (method =="REGISTER")
record_route();
# loose-route processing
if (loose_route()) {
#commented 11/02/05
#t_relay();
route(1);
break;
};
# 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 (uri==myself) {
log(1,"into loop");
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("157.190.74.152", "subscriber")) {
# www_challenge("157.190.74.152", "0");
# break;
# };
#cpl_process_register();
save("location");
break;
};
# if (method=="SUBSCRIBE")
&g! t; # {
# log(1, "Subscribe\n");
# if(t_newtran())
# {
# log(1, "Registrar\n");
# handle_subscription("registrar");
# };
# break;
# };
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
if (method=="INVITE"){
log(1,"in ! invite loop");
#break; #no 100 trying
#if (!cpl_run_script("incoming","is_stateless"))
#{
# #script execution failed
# t_reply("500", "CPL script execution failed");
#};
t_on_failure("1");
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
#sl_send_reply("404", "Not Found");
route(2);
break;
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
#commented 11/02/05#######################
if (!t_relay()) {
sl_reply_error();
};
#route(1);
}
######################################entered
11/02/05############################################################
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");
break;
};
if (isflagset(6)){
force_rtp_proxy();
}
t_on_reply("1");
if(!t_relay()){
sl_reply_error();
break;
};
}
######################################entered
11/02/05############################################################
#!! Nathelper
onreply_route[1] {
if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){
fix_nated_contact();
force_rtp_proxy();
}else if (nat_uac_test("1")){
fix_nated_contact();
};
}
#################################################################################################################
# ------------- handling of unavai! lable user ------------------
route[2] {
# non-Voip -- just send "off-line"
if (!(method == "INVITE" ! || method == "ACK" || method ==
"CANCEL")) {
sl_send_reply("404", "Not Found");
break;
};
# forward to voicemail now
rewritehostport("157.190.74.152:5062");
t_relay_to_udp("157.190.74.152", "5062");
}
# if forwarding downstream did not succeed, try voicemail running
# at 172.16.2.120:5062
failure_route[1] {
revert_uri();
rewritehostport("157.190.74.152:5062");
append_branch();
t_relay_to_udp("157.190.74.152", "5062");
}
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Send instant messages to your
online friends
http://uk.messenger.yahoo.com
7177
days inactive
7178
days old
6 comments
2 participants
participants (2)
-
Greger V. Teigre -
Vivienne Curran