[Kamailio-Users] No Audio with clients Behind NAT, audio is fine with clients using public IPs (Im using rtpproxy)

rubenrojas - Trc.es rubenrojas at trc.es
Thu Jun 25 12:41:11 CEST 2009


Hello everyone, this is my first post on this list,

I have installed kamailio 1.5.1 and set up a vanilla default kamailio.cfg, then I have modified the cfg to activate mysql, domain, presence, nathelper and authentication with md5, everything works as supposed to, and the clients can register, send txt messages and talk to each other. The only problem is with the audio when the two clients are behind a NAT, the phones can make a call and it does ring too, but when you pick up there is no audio both ways.

when the phones have a public IP everything goes fine, it also works when I use a Linksys PAP2T whith the options to "Insert VIA received", "Insert VIA rport", "Handle VIA received", "Handle VIA rport" and "NAT mapping enable" turned on, with the Qutecom softphone works too.

This is happening with thomson phones (model ST 2022), and GrandStream Budge Tone 200, it happens no matter what options I set for NATting on the phones, I've even used stun with stunserver.org or the ekiga stunserver, the phones register and can make and recieve calls, but there is no audio when you pick up the call.

With a kamctl ul show, you can see that the phones have registered the Contact with their local IPs and the Received have the public IPs and ports for the NAT
The only difference with the working Linksys is that they register the Contact with the public IP.
Here you can see two NATed phones on the proxy

Domain:: location table=512 records=2 max_slot=1
        AOR:: 20000004 at 212.4.107.250
                Contact:: sip:20000004 at 192.168.254.110:5060;transport=udp;user=phone Q=
                        Expires:: 1150
                        Callid:: 72ed03f6d2f390f9 at 192.168.254.110
                        Cseq:: 10003
                        User-agent:: Grandstream BT200 1.1.6.27
                        Received:: sip:212.4.97.115:35379
                        State:: CS_NEW
                        Flags:: 0
                        Cflag:: 0
                        Socket:: udp:212.4.107.250:5060
                        Methods:: 7807
        AOR:: 20000000 at 212.4.107.250
                Contact:: sip:20000000 at 192.168.254.101:5060;user=phone Q=
                        Expires:: 2945
                        Callid:: 17fe-c0a80101-5-1 at 192.168.254.101
                        Cseq:: 6
                        User-agent:: THOMSON ST2022 hw2 fw3.56 00-18-F6-B5-7E-06
                        Received:: sip:212.4.97.115:55128
                        State:: CS_NEW
                        Flags:: 0
                        Cflag:: 0
                        Socket:: udp:212.4.107.250:5060
                        Methods:: 4294967295


Im using rtpproxy and there is no log error that indicates that rttpproxy isn't working, in fact doing a SIP trace shows rtpproxy setting ports for the audio.
I run rtpproxy with this command:

rtpproxy -l 212.4.107.250 -s udp:localhost:7722 -F

Any help would be greatly appreciated, I've been two weeks looking for a solution

Im attaching my kamailio.cfg so you can take a look, at the end of the message Im gonna attache the SIP Trace of a call between two NATed phones (a Thomson and a GrandStream) in case anyone can help me decypher whats wrong here:

this is my cfg file
**************************************************************************************************

#
# $Id: kamailio.cfg 5800 2009-04-20 11:01:49Z miconda $
#
# Kamailio (OpenSER) SIP Server - basic configuration script
#     - web: http://www.kamailio.org
#     - svn: http://openser.svn.sourceforge.net/viewvc/openser/
#
# Direct your questions about this file to: <users at 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.
#
# There are comments showing how to enable different features in th econfig
# file. Such commented code starts with #X# where X is a letter to identify
# a feature. Delete entire #X# if you want to enable that feature. Next are
# sed commands that help you enable such features.
#
# *** To enamble mysql execute:
#     sed -i 's/#m#//g' kamailio.cfg
#
# *** To enamble authentication execute:
#     - enable mysql
#     sed -i 's/#a#//g' kamailio.cfg
#     - add users using 'kamctl'
#
# *** To enamble persistent user location execute:
#     - enable mysql
#     sed -i 's/#u#//g' kamailio.cfg
#
# *** To enamble presence server execute:
#     - enable mysql
#     sed -i 's/#p#//g' kamailio.cfg
#
# *** To enamble nat traversal execute:
#     sed -i 's/#n#//g' kamailio.cfg
#     - install RTPProxy: http://www.rtpproxy.org
#     - start RTPProxy:
#        rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#
# *** To enhance accounting execute:
#     - enable mysql
#     sed -i 's/#c#//g' kamailio.cfg
#     - add following columns to database
# 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_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
# ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
#


####### Global Parameters #########

debug=3
log_stderror=no
log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes

/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes

/* uncomment the next line to enable the auto temporary blacklisting of 
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns 
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* uncomment the next line to disable the auto discovery of local aliases
   based on revers DNS on IPs (default on) */
#auto_aliases=no

/* uncomment the following lines to enable TLS support  (default off) */
#disable_tls = no
#listen = tls:your_IP:5061
#tls_verify_server = 1
#tls_verify_client = 1
#tls_require_client_certificate = 0
#tls_method = TLSv1
#tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem"
#tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem"
#tls_ca_list     = "/usr/local/etc/kamailio/tls/user/user-calist.pem"


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:192.168.1.2:5060


####### Modules Section ########

#set module path
mpath="/usr/local/lib/kamailio/modules/"

/* uncomment next line for MySQL DB support */
loadmodule "db_mysql.so"
loadmodule "mi_fifo.so"
loadmodule "sl.so"
loadmodule "tm.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 "acc.so"
/* uncomment next lines for MySQL based authentication support 
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "auth.so"
loadmodule "auth_db.so"
/* 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"
/* uncomment the next two lines for presence server support
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "presence.so"
loadmodule "presence_xml.so"
loadmodule "presence_mwi.so"#manually added

loadmodule "nathelper.so"

# ----------------- 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=$rd")
/* uncomment the following lines to enable DB accounting also */
#c#modparam("acc", "db_flag", 1)
#c#modparam("acc", "db_missed_flag", 2)
#c#modparam("domain", "db_url",
#c#	"mysql://openser:openserrw@localhost/openser")
#c#modparam("acc", "db_extra",
#c#	"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")


# ----- usrloc params -----
/* uncomment the following lines if you want to enable DB persistency
   for location entries */
#u#modparam("usrloc", "db_mode",   2)
#u#modparam("usrloc", "db_url",
#u#	"mysql://openser:openserrw@localhost/openser")

# ----- auth_db params -----
/* uncomment the following lines if you want to enable the DB based
   authentication */
#a#modparam("auth_db", "calculate_ha1", yes)
#a#modparam("auth_db", "password_column", "password")
#a#modparam("auth_db", "db_url",
#a#	"mysql://openser:openserrw@localhost/openser")
#a#modparam("auth_db", "load_credentials", "")

#parametros de autentificacion modificados manualmente
modparam("auth_db", "user_column", "username")
modparam("auth_db", "domain_column", "domain")
modparam("auth_db", "password_column", "ha1")
modparam("auth_db", "password_column_2", "ha1b")
modparam("auth_db", "calculate_ha1", 0)
#modparam("auth_db", "use_domain", 0)
modparam("auth_db", "use_domain", 1)#0 encendemos con 1 porque utilizaremos multi-domain
modparam("auth_db", "load_credentials", "rpid")
modparam("auth_db", "db_url",
	"mysql://openser:openserrw@localhost/openser")


# ----- alias_db params -----
/* uncomment the following lines if you want to enable the DB based
   aliases */
#modparam("alias_db", "db_url",
#	"mysql://openser:openserrw@localhost/openser")


# ----- domain params -----
/* uncomment the following lines to enable multi-domain detection
   support */
modparam("domain", "db_url",
	"mysql://openser:openserrw@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 -----
/* uncomment the following lines if you want to enable presence */
modparam("presence|presence_xml", "db_url",
	"mysql://openser:openserrw@localhost/openser")
modparam("presence_xml", "force_active", 1)
modparam("presence", "server_address", "sip:212.4.107.250:5060")

# -- nathelper
modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722")
modparam("nathelper", "natping_interval", 15)
modparam("nathelper", "ping_nated_only", 0)
modparam("nathelper", "sipping_bflag", 7)
modparam("nathelper", "sipping_from", "sip:pinger at 212.4.107.250")
modparam("registrar|nathelper", "received_avp", "$avp(i:80)")
modparam("usrloc", "nat_bflag", 6)
modparam("nathelper", "sipping_method", "OPTIONS")


####### Routing Logic ########


# main request routing logic

route{

	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		exit;
	}

	# NAT detection
	route(4);

	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(1);
		} else {
			if (is_method("SUBSCRIBE") && uri == myself) {
				# in-dialog subscribe requests
				route(2);
				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(3);

	# record routing
	if (!is_method("REGISTER|MESSAGE"))
	{
		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"); 
		# if you have some interdomain connections via TLS
		##if($rd=="tls_domain1.net") {
		##	t_relay("tls:domain1.net");
		##	exit;
		##} else if($rd=="tls_domain2.net") {
		##	t_relay("tls:domain2.net");
		##	exit;
		##}
		route(1);
	}

	# requests for my domain

	if( is_method("PUBLISH|SUBSCRIBE"))
	{
		route(2);
	}

	if (is_method("REGISTER"))
	{
		if (!save("location"))
		{
			sl_reply_error();
		}
		exit;
	}

	if ($rU==NULL) {
		# request with no Username in RURI
		sl_send_reply("484","Address Incomplete");
		exit;
	}

	# apply DB based aliases (uncomment to enable)
	##alias_db_lookup("dbaliases");

	if (!lookup("location")) {
		switch ($retcode) {
			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(1);
}


route[1] {
	if (check_route_param("nat=yes")) {
		setbflag(6);
		setbflag(7);# sipping
	}
	if (isflagset(5) || isbflagset(6)) {
		route(5);
	}

	/* example how to enable some additional event routes */
	if (is_method("INVITE")) {
		#t_on_branch("1");
		t_on_reply("1");
		t_on_failure("1");
	}

	if (!t_relay()) {
		sl_reply_error();
	}
	exit;
}


# Presence route
/* uncomment the whole following route for enabling presence server */
route[2]
{
	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;
	
	# 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
/* uncomment the whole following route for enabling authentication */
route[3] {
	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;
		}
	}
# Auth only on registration
#a#	} else {
#a#		# authenticate if from local subscriber (uncomment to enable auth)
#a#		if (from_uri==myself)
#a#		{
#a#			if (!proxy_authorize("", "subscriber")) {
#a#				proxy_challenge("", "0");
#a#				exit;
#a#			}
#a#			if (is_method("PUBLISH"))
#a#			{
#a#				if ($au!=$tU) {
#a#					sl_send_reply("403","Forbidden auth ID");
#a#					exit;
#a#				}
#a#			} else {
#a#				if ($au!=$fU) {
#a#					sl_send_reply("403","Forbidden auth ID");
#a#					exit;
#a#				}
#a#			}
#a#
#a#			consume_credentials();
#a#			# caller authenticated
#a#		}
#a#	}
	return;
}

# Caller NAT detection route
/* uncomment the whole following route for enabling Caller NAT Detection */
route[4]{
	force_rport();
	if (nat_uac_test("19")) {
		if (method=="REGISTER") {
			fix_nated_register();
		} else {
			fix_nated_contact();
		}
		setflag(5);
	}
	return;
}

# RTPProxy control
/* uncomment the whole following route for enabling RTPProxy Control */
route[5] {
	if (is_method("BYE")) {
		unforce_rtp_proxy();
	} else if (is_method("INVITE")){
		force_rtp_proxy();
	}
	if (!has_totag()) add_rr_param(";nat=yes");
	return;
}

branch_route[1] {
	xdbg("new branch at $ru\n");
}


onreply_route[1] {
	xdbg("incoming reply\n");

	if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") {
		force_rtp_proxy();
	}
	if (isbflagset(6)) {
		fix_nated_contact();
	}
}


failure_route[1] {
	if (is_method("INVITE")
			&& (isbflagset(6) || isflagset(5))) {
		unforce_rtp_proxy();
	}

	if (t_was_cancelled()) {
		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();
	##}
}

**************************************************************************************************
**************************************************************************************************

And here goes the SIP Trace for a NATed to NATed hardphones:
**************************************************************************************************
U +0.161561 212.4.97.115:35379 -> 212.4.107.250:5060
INVITE sip:20000000 at 212.4.107.250;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668
From: "20000004" <sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>
Contact: <sip:20000004 at 192.168.254.110:5060;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 332

v=0
o=20000004 8000 8000 IN IP4 192.168.254.110
s=SIP Call
c=IN IP4 192.168.254.110
t=0 0
m=audio 40000 RTP/AVP 4 3 18 0 8 9 97
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:60

#
U +0.000407 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668;rport=35379;received=212.4.97.115
From: "20000004" <sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
Server: Kamailio (1.5.1-notls (i386/linux))
Content-Length: 0


#
U +0.000034 212.4.107.250:5060 -> 212.4.97.115:55128
INVITE sip:20000000 at 192.168.254.101:5060;user=phone SIP/2.0
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004" <sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>
Contact: <sip:20000004 at 212.4.97.115:35379;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 348

v=0
o=20000004 8000 8000 IN IP4 192.168.254.110
s=SIP Call
c=IN IP4 212.4.107.250
t=0 0
m=audio 35752 RTP/AVP 4 3 18 0 8 9 97
a=sendrecv
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:60
a=nortpproxy:yes

#
U +0.019311 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004"<sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
Content-Length: 0


#
U +0.030480 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004"<sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000 at 192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Length: 0


#
U +0.000083 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004"<sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000 at 192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Length: 0


#
U +6.510103 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004"<sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000 at 192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151

v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv

#
U +0.000365 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004"<sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000 at 192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167

v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes

#
U +0.034122 212.4.97.115:35379 -> 212.4.107.250:5060
ACK sip:20000000 at 192.168.254.101:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bKdf5e0ceed72f3797
Route: <sip:212.4.107.250;lr=on;nat=yes>
From: "20000004" <sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>;tag=c0a80101-21188
Contact: <sip:20000004 at 192.168.254.110:5060;transport=udp;user=phone>
Supported: path
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 ACK
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0


#
U +0.000245 212.4.107.250:5060 -> 192.168.254.101:5060
ACK sip:20000000 at 192.168.254.101:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.2
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bKdf5e0ceed72f3797
From: "20000004" <sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>;tag=c0a80101-21188
Contact: <sip:20000004 at 212.4.97.115:35379;transport=udp;user=phone>
Supported: path
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 ACK
User-Agent: Grandstream BT200 1.1.6.27
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0


#
U +0.458031 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004"<sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000 at 192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151

v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv

#
U +0.000246 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004"<sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000 at 192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167

v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes

#
U +0.999724 212.4.97.115:55128 -> 212.4.107.250:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004"<sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000 at 192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 151

v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 192.168.254.101
t=0 0
m=audio 32448 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv

#
U +0.000295 212.4.107.250:5060 -> 212.4.97.115:35379
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668
From: "20000004"<sip:20000004 at 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
To: <sip:20000000 at 212.4.107.250;user=phone>;tag=c0a80101-21188
Call-ID: c177cae013da224d at 192.168.254.110
CSeq: 29653 INVITE
Require: timer
Session-Expires: 100;refresher=uac
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20000000 at 192.168.254.101:5060;user=phone>
Record-Route: <sip:212.4.107.250;lr=on;nat=yes>
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 167

v=0
o=20000000 138812 138812 IN IP4 192.168.254.101
s=-
c=IN IP4 212.4.107.250
t=0 0
m=audio 35754 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=nortpproxy:yes





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