Hi there,
I am pretty new to all this (kamailio..)
I am trying to set up the following:
1. Kamailio with web-gui -> done
2. Asterisk / elastix for connecting to sip-provider / PSTN-> done
This works perfectly incoming / outgoing
3. Connecting elastix/asterix to kamailio -> done
I already have extensions configured on kamailio. They can call each other
perfectly.
Now I do need a route-config or dialplan that routes ALL outgoing calls for
all extensions to the
Extension of the asterisk/elastix-machine.
Is that possible and if so, how ?
I do need to provision more than a few hundred sip-subscribers in the near
future so connecting
Them using just asterisk is not an option..
Thanks in advance, I really appreciate your help
Regards
lutz
hi all,
I am using Twinkle on fedora 11 as a client. it connected to the server
fine but after some time it started giving on register as well as
de-register.
[Jun 18 15:15:14] ERROR[14964]: chan_sip.c:12215 register_verify: 'UDP'
is not a valid transport for '70003'. we only use 'UNKNOWN'! ending
call.
[Jun 18 15:15:14] NOTICE[14964]: chan_sip.c:20561
handle_request_register: Registration from '"70003" <sip:70003@a.a.a.a>'
failed for 'x.x.x.x' - Device not configured to use this transport type
what could be the reason and the solution?
regards
Siddharth
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2010/6/18 Stefan Sayer <stefan.sayer(a)googlemail.com>:
> but is that also true if the "Digest" appears as is in the ABNF?
>
> just curious
Yes, there is no special reason for it.
RFC 3261 - 25.1
-------------------------
Many SIP header field values consist of words separated by LWS or
special characters. Unless otherwise stated, tokens are case-
insensitive.
-------------------------
As all we know the schema in a SIP/SIPS URI is case insensitive
("sip:" == "SIP:"):
SIP-URI = "sip:" [ userinfo ] hostport
uri-parameters [ headers ]
However SIP methods are case sensitive ("INVITE" != "invite"):
INVITEm = %x49.4E.56.49.54.45 ; INVITE in caps
So *any* string appearing in SIP BNF (as "Digest", "MD5", "tag"....)
are case insensitive.
PS:
/Ragel-SIP-Parser/sip_parser.sh Authorization 'Authorization: DIGEST
username="Alice", realm="atlanta.com",
nonce="84a4cc6f3082121f32b42a2187831a9e",
response="7587245234b3434cc3412213e5f113a5"'
OK
;)
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
Greetings,
When I use uac_auth() in a failure route in this manner:
if(t_check_status("407")) {
uac_auth();
append_branch();
t_relay();
}
or the reverse order vis-a-vis uac_auth() and append_branch(), I do not
see any digest authentication headers being inserted into the
retransmitted INVITE request. Also, CSeq is not being incremented.
This is Kamailio 3.0.2.
Please advise.
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
> Date: Tue, 15 Jun 2010 16:57:37 -0400
> From: Alex Balashov <abalashov(a)evaristesys.com>
> Subject: Re: [SR-Users] sipp to kamailio doesn't match ACK
> To: sr-users(a)lists.sip-router.org
> Message-ID: <4C17E941.4020506(a)evaristesys.com>
> Content-Type: text/plain; charset=UTF-8; format=flowed
>
> 1) AFAIK, the Record-Route/Route headers are used by TM for maintaining
> transaction state, unless you are relying on the 'dialog' module's "fast
> matching" cookies, which are identifiers that get placed as a URI
> parameter into the Record-Route header so that it is not necessary to
> match the message by more conventional means (Via/branch, Call-ID, tags,
> etc.)
>
> 2) ACK is a separate transaction; every request and reply sequence is a
> transaction. So, if you're trying to match it to an existing INVITE
> transaction, for instance, that won't happen.
>
> t_check_trans() has special behaviour for ACKs, depending on whether
> they're end-to-end ACKs or ACKs for negative replies to existing
> transactions:
>
> http://www.kamailio.org/docs/modules/3.0.x/modules/tm.html#t_check_trans
>
> Are you sure that you're using it, or its newer variant,
> t_lookup_request(), correctly?
>
> 3) You are, nevertheless, correct that the ACK should contain the RR
> header:
>
> The UAC core MUST generate an ACK request for each 2xx received from
> the transaction layer. The header fields of the ACK are constructed
> in the same way as for any request sent within a dialog (see Section
> 12) with the exception of the CSeq and the header fields related to
> authentication.
>
> -- Alex
>
Thanks for the feedback Alex.
JR
Hi All,
I'm building a video conference system with the kamailio and the video AS. The network is as below:
UAC(behind NAT) ------ kamailio(public IP) ------ AS (public IP)
I have setup the basic kamailio and the SIP signaling can reach the AS and the call can be established. Now the time to setup the NAT traveral since the UACs are behind the NAT.
Since it's video scenario, I don't want the kamailio as the RTPproxy so try to use the NAT_Traversal module.
>From the module doc, NAT_Traversal needs the Dialog module as well. Have no idea on how to load these two modules and the route config required although have readed the module docs. Wondering if someone can help on the config for this scenario?
All call with $rU=="0216666" is forwarded to the video AS as the route[CONF].
Thanks in advance!
Thanks,
Kevin
Below is the current config of the kamailio server:
==============CFG=============
AppSer01:root@/usr/local/kamailio-3.0/etc/kamailio$ cat kamailio.cfg
#!KAMAILIO
#!define WITH_DEBUG
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
# $Id$
#
# Kamailio (OpenSER) SIP Server v3.0 - basic configuration script
# - web: http://www.kamailio.org
# - git: http://sip-router.org
#
# Direct your questions about this file to: <users(a)lists.kamailio.org>
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode:
# - define WITH_DEBUG
#
# *** To enable mysql:
# - define WITH_MYSQL
#
# *** To enable authentication execute:
# - enable mysql
# - define WITH_AUTH
# - add users using 'kamctl'
#
# *** To enable persistent user location execute:
# - enable mysql
# - define WITH_USRLOCDB
#
# *** To enable presence server execute:
# - enable mysql
# - define WITH_PRESENCE
#
# *** To enable nat traversal execute:
# - define WITH_NAT
# - install RTPProxy: http://www.rtpproxy.org
# - start RTPProxy:
# rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#
# *** To enable PSTN gateway routing execute:
# - define WITH_PSTN
# - set the value of pstn.gw_ip
# - check route[PSTN] for regexp routing condition
#
# *** To enhance accounting execute:
# - enable mysql
# - define WITH_ACCDB
# - add following columns to database
#!ifdef ACCDB_COMMENT
ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
#!endif
####### Global Parameters #########
#!ifdef WITH_DEBUG
debug=4
log_stderror=no
#!else
debug=2
log_stderror=no
#!endif
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
fork=yes
children=4
/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases
based on revers DNS on IPs (default on) */
#auto_aliases=no
port=5060
/* uncomment and configure the following line if you want Kamailio to
bind on a specific interface/port/proto (default bind on all available) */
listen=udp:210.13.124.15:5060
####### Custom Parameters #########
# These parameters can be modified runtime via RPC interface
# - see the documentation of 'cfg_rpc' module.
#
# Format: group.id = value 'desc' description
# Access: $sel(cfg_get.group.id) or @cfg_get.group.id
#
#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"
#
# - by default is empty to avoid misrouting
pstn.gw_ip = "" desc "PSTN GW Address"
#!endif
####### Modules Section ########
#set module path
mpath="/usr/local/kamailio-3.0/lib/kamailio/modules_k/:/usr/local/kamailio-3.0/lib/kamailio/modules/"
/* uncomment next line for MySQL DB support */
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "uri_db.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!endif
/* uncomment next line for aliases support
NOTE: a DB (like db_mysql) module must be also loaded */
#loadmodule "alias_db.so"
/* uncomment next line for multi-domain support
NOTE: a DB (like db_mysql) module must be also loaded
NOTE: be sure and enable multi-domain support in all used modules
(see "multi-module params" section ) */
#loadmodule "domain.so"
#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
#!endif
#!ifdef WITH_NAT
loadmodule "nathelper.so"
#!endif
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)
# ----- rr params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
# ----- uri_db params -----
/* by default we disable the DB support in the module as we do not need it
in this configuration */
modparam("uri_db", "use_uri_table", 0)
modparam("uri_db", "db_url", "")
# ----- acc params -----
/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
/* by default ww do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
/* enhanced DB accounting */
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("acc", "db_url",
"mysql://openser:openK2mrw23@localhost/openser")
modparam("acc", "db_extra",
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
#!endif
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url",
"mysql://openser:openK2mrw23@localhost/openser")
#!endif
# ----- auth_db params -----
/* enable the DB based authentication */
#!ifdef WITH_AUTH
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",
"mysql://openser:openK2mrw23@localhost/openser")
modparam("auth_db", "load_credentials", "")
#!endif
# ----- alias_db params -----
/* uncomment the following lines if you want to enable the DB based
aliases */
#modparam("alias_db", "db_url",
# "mysql://openser:openK2mrw23@localhost/openser")
# ----- domain params -----
/* uncomment the following lines to enable multi-domain detection
support */
#modparam("domain", "db_url",
# "mysql://openser:openK2mrw23@localhost/openser")
#modparam("domain", "db_mode", 1) # Use caching
# ----- multi-module params -----
/* uncomment the following line if you want to enable multi-domain support
in the modules (dafault off) */
#modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1)
# ----- presence params -----
/* enable presence server support */
#!ifdef WITH_PRESENCE
modparam("presence|presence_xml", "db_url",
"mysql://openser:openK2mrw23@localhost/openser")
modparam("presence_xml", "force_active", 1)
modparam("presence", "server_address", "sip:10.0.0.10:5060")
#!endif
# ----- nathelper -----
#!ifdef WITH_NAT
modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722")
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", 7)
modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org")
modparam("registrar|nathelper", "received_avp", "$avp(i:80)")
modparam("usrloc", "nat_bflag", 6)
#!endif
####### Routing Logic ########
# main request routing logic
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if(!sanity_check("1511", "7"))
{
xlog("Malformed SIP message from $si:$sp\n");
exit;
}
# NAT detection
route(NAT);
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
}
route(RELAY);
} else {
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ... ignore and discard.\n");
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
#initial requests
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
# account only INVITEs
if (is_method("INVITE")) {
setflag(1); # do accounting
}
if (!uri==myself)
/* replace with following line if multi-domain support is used */
##if (!is_uri_host_local())
{
append_hf("P-hint: outbound\r\n");
route(RELAY);
}
# requests for my domain
if( is_method("PUBLISH|SUBSCRIBE"))
route(PRESENCE);
if (is_method("REGISTER"))
{
if(isflagset(5))
{
setbflag("6");
# uncomment next line to do SIP NAT pinging
## setbflag("7");
}
if (!save("location"))
sl_reply_error();
exit;
}
if ($rU==$null) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
route(PSTN);
route(CONF);
# apply DB based aliases (uncomment to enable)
##alias_db_lookup("dbaliases");
if (!lookup("location")) {
switch ($rc) {
case -1:
case -3:
t_newtran();
t_reply("404", "Not Found");
exit;
case -2:
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
route(RELAY);
}
route[RELAY] {
#!ifdef WITH_NAT
if (check_route_param("nat=yes")) {
setbflag("6");
}
if (isflagset(5) || isbflagset("6")) {
route(RTPPROXY);
}
#!endif
/* example how to enable some additional event routes */
if (is_method("INVITE")) {
#t_on_branch("BRANCH_ONE");
t_on_reply("REPLY_ONE");
t_on_failure("FAIL_ONE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Presence server route
route[PRESENCE]
{
#!ifdef WITH_PRESENCE
if (!t_newtran())
{
sl_reply_error();
exit;
};
if(is_method("PUBLISH"))
{
handle_publish();
t_release();
}
else
if( is_method("SUBSCRIBE"))
{
handle_subscribe();
t_release();
}
exit;
#!endif
# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==$null)
{
sl_send_reply("404", "Not here");
exit;
}
return;
}
# Authentication route
route[AUTH] {
#!ifdef WITH_AUTH
if (is_method("REGISTER"))
{
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize("", "subscriber"))
{
www_challenge("", "0");
exit;
}
if ($au!=$tU)
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
} else {
# authenticate if from local subscriber (uncomment to enable auth)
if (from_uri==myself)
{
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (is_method("PUBLISH"))
{
if ($au!=$tU) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
} else {
if ($au!=$fU) {
sl_send_reply("403","Forbidden auth ID");
exit;
}
}
consume_credentials();
# caller authenticated
}
}
#!endif
return;
}
# Caller NAT detection route
route[NAT]{
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (method=="REGISTER") {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(5);
}
#!endif
return;
}
# RTPProxy control
route[RTPPROXY] {
#!ifdef WITH_NAT
if (is_method("BYE")) {
unforce_rtp_proxy();
} else if (is_method("INVITE")){
force_rtp_proxy();
}
if (!has_totag()) add_rr_param(";nat=yes");
#!endif
return;
}
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
# check if PSTN GW IP is defined
if (strempty($sel(cfg_get.pstn.gw_ip))) {
xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");
return;
}
# route to PSTN dialed numbers starting with '+' or '00'
# (international format)
# - update the condition to match your dialing rules for PSTN routing
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
return;
# only local users allowed to call
if(from_uri!=myself) {
sl_send_reply("403", "Not Allowed");
exit;
}
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
route(RELAY);
exit;
#!endif
return;
}
# add by KJ --start
route[CONF] {
if(is_method("INVITE") && $rU=="0216666") {
rewritehostport("210.x.y.z:5060");
}
route(RELAY);
exit;
}
# add by KJ --end
# Sample branch router
branch_route[BRANCH_ONE] {
xdbg("new branch at $ru\n");
}
# Sample onreply route
onreply_route[REPLY_ONE] {
xdbg("incoming reply\n");
#!ifdef WITH_NAT
if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") {
force_rtp_proxy();
}
if (isbflagset("6")) {
fix_nated_contact();
}
#!endif
}
# Sample failure route
failure_route[FAIL_ONE] {
#!ifdef WITH_NAT
if (is_method("INVITE")
&& (isbflagset("6") || isflagset(5))) {
unforce_rtp_proxy();
}
#!endif
if (t_is_canceled()) {
exit;
}
# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status("3[0-9][0-9]")) {
##t_reply("404","Not found");
## exit;
##}
# uncomment the following lines if you want to redirect the failed
# calls to a different new destination
##if (t_check_status("486|408")) {
## sethostport("192.168.2.100:5060");
## append_branch();
## # do not set the missed call flag again
## t_relay();
##}
}
AppSer01:root@/usr/local/kamailio-3.0/etc/kamailio$
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On Tue, Jun 15, 2010 at 3:44 PM, JR Richardson <jmr.richardson(a)gmail.com> wrote:
> Hi All,
>
> I have been in the lab testing various load balancing scenarios with
> kamailio 1.5 and 3.0 and distributing calls to asterisk servers. I am
> running into an issue that I'm sure is related to sipp not kamailio or
> asterisk. My testing is setup like this:
> sipp><kamailio stateful (dispatcher or carrierroute)><multiple asterisk servers
>
> My sipp scenario:
> sipp -sn uac 10.10.12.22 -i 10.10.14.97 -s 22265 -d 5000 -l 100 -r 1
> -trace_err -m 1 -nr
>
> What I'm seeing in my ngrep sip traces is the ACK coming back from
> sipp does not have the RR header or a proper ftag:
>
> Here is the Ok sent from kamailio 10.10.12.22 to sipp 10.10.14.97:
>
> U 10.10.12.22:5060 -> 10.10.14.97:5061
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-0.
> Record-Route: <sip:10.10.12.22;lr=on;ftag=10002SIPpTag001>.
> From: sipp <sip:sipp@10.10.14.97:5061>;tag=10002SIPpTag001.
> To: sut <sip:22265@10.10.12.22:5060>;tag=as18ae7aa7.
> Call-ID: 1-10002(a)10.10.14.97.
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: <sip:65@10.10.14.102>.
> Content-Type: application/sdp.
> Content-Length: 184.
> .
> v=0.
> o=root 23641 23641 IN IP4 10.10.14.102.
> s=session.
> c=IN IP4 10.10.14.102.
> t=0 0.
> m=audio 24270 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
>
> Here is the reply ACK sent from sipp 10.10.14.97 to kamailio 10.10.12.22:
>
> U 10.10.14.97:5061 -> 10.10.12.22:5060
> ACK sip:22265@10.10.12.22:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-5.
> From: sipp <sip:sipp@10.10.14.97:5061>;tag=10002SIPpTag001.
> To: sut <sip:22265@10.10.12.22:5060>;tag=as18ae7aa7.
> Call-ID: 1-10002(a)10.10.14.97.
> CSeq: 1 ACK.
> Contact: sip:sipp@10.10.14.97:5061.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
>
> And the ACK goes into the to-tag portion of the kamailio route script
> but can not match the transaction so kamailio exits, the dialog drops
> out of memory and when sipp sends a BYE, kamailio replys with 404 not
> found.
>
> So if I'm diagnosing this correctly, sipp is not maintaining the
> Record-Route: header in the responses back to kamailio and without
> that info kamailio can not maintain transaction state and the call
> fails. Is there any work around or possibly another SIP performance
> testing suite that will handle RR headers?
>
> Thanks.
>
> JR
> --
> JR Richardson
> Engineering for the Masses
>
After reassuring myself that my diagnosis was correct and shortly
before I went blind tracing sip messages, I dug into sipp and found
the ability to create a new scenario adding the record route header
and next_url parameter. So now when I run the testing from sipp, the
dialogs are properly processed statefully through kamailio and all
calls complete as expected.
I though I new SIP pretty well, but it's the problems I run into that
convince me there is so much more to learn.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
Hi All,
I have been in the lab testing various load balancing scenarios with
kamailio 1.5 and 3.0 and distributing calls to asterisk servers. I am
running into an issue that I'm sure is related to sipp not kamailio or
asterisk. My testing is setup like this:
sipp><kamailio stateful (dispatcher or carrierroute)><multiple asterisk servers
My sipp scenario:
sipp -sn uac 10.10.12.22 -i 10.10.14.97 -s 22265 -d 5000 -l 100 -r 1
-trace_err -m 1 -nr
What I'm seeing in my ngrep sip traces is the ACK coming back from
sipp does not have the RR header or a proper ftag:
Here is the Ok sent from kamailio 10.10.12.22 to sipp 10.10.14.97:
U 10.10.12.22:5060 -> 10.10.14.97:5061
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-0.
Record-Route: <sip:10.10.12.22;lr=on;ftag=10002SIPpTag001>.
From: sipp <sip:sipp@10.10.14.97:5061>;tag=10002SIPpTag001.
To: sut <sip:22265@10.10.12.22:5060>;tag=as18ae7aa7.
Call-ID: 1-10002(a)10.10.14.97.
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: <sip:65@10.10.14.102>.
Content-Type: application/sdp.
Content-Length: 184.
.
v=0.
o=root 23641 23641 IN IP4 10.10.14.102.
s=session.
c=IN IP4 10.10.14.102.
t=0 0.
m=audio 24270 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
Here is the reply ACK sent from sipp 10.10.14.97 to kamailio 10.10.12.22:
U 10.10.14.97:5061 -> 10.10.12.22:5060
ACK sip:22265@10.10.12.22:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-5.
From: sipp <sip:sipp@10.10.14.97:5061>;tag=10002SIPpTag001.
To: sut <sip:22265@10.10.12.22:5060>;tag=as18ae7aa7.
Call-ID: 1-10002(a)10.10.14.97.
CSeq: 1 ACK.
Contact: sip:sipp@10.10.14.97:5061.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
And the ACK goes into the to-tag portion of the kamailio route script
but can not match the transaction so kamailio exits, the dialog drops
out of memory and when sipp sends a BYE, kamailio replys with 404 not
found.
So if I'm diagnosing this correctly, sipp is not maintaining the
Record-Route: header in the responses back to kamailio and without
that info kamailio can not maintain transaction state and the call
fails. Is there any work around or possibly another SIP performance
testing suite that will handle RR headers?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
Hi
I have build SIP-Router and modules from sources from GIT repository
(development branch).
auth_radius module results in a "Segmentation fault: 11" when I try to
register (radius_www_authorize).
sercmd> core.version
Server: ser (3.0.99-dev1 (i386/freebsd)) eabca5
Any help for solving the problem would be appreciated. Thanks.
--
Dmitry Kirillov, System Engineer
KDS-RIPE
Network Operational Center, JSC Neotelecom
Tomsk, Russia
Telephone: +7(3822)710-008, ICQ: 22413725
Web: http://www.neotelecom.ru/ | mailto:noc@neotelecom.ru
Looking Glass: http://lg.neotelecom.ru/
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