[SR-Users-ES] No Audio con clientes detras de una NAT, el audio funciona con clientes que usan IPs publicas (Estoy utilizando rtpproxy)
ggb
ggb en tid.es
Jue Jun 25 13:17:04 CEST 2009
En el INVITE reenviado de Kamailio al destino se ve la IP privada en el
SDP.
Tal vez no esté entrando en el route[5] que es donde se hace el
force_rtp_proxy. Prueba a que se ejecute siempre el force_rtp_proxy en
los INVITEs y sus respuestas a ver si así te funciona en este caso que
te da problemas.
G.
On 06/25/2009 01:01 PM, rubenrojas - Trc.es wrote:
> Hola, este es mi primer post en esta lista,
>
>
> Tengo instalado el kamailio 1.5.1 y estoy utilizando un kamailio.cfg utilizando el que trae por defecto, luego he modificado el cfg para activar mysql, domain, presence, nathelper y authentication con md5, todo funciona como se supone que deberia, los clientes pueden registrarse, enviar mensajes de texto y hablar entre ellos. El unico problema es el audio cuando dos clientes estan detras de una NAT, los telefonos pueden realizar la llamada y suena el ring, pero cuando se descuelga no hay audio en ninguna direccion.
>
> cuendo los telefonos tienen una IP publica todo funciona bien, tambien funciona cuando utilizo un Linksys PAP2T con las opciones "Insert VIA received", "Insert VIA rport", "Handle VIA received", "Handle VIA rport" y "NAT mapping enable" encendidas, con el softphone de Qutecom stambien funciona.
>
> Este problema me esta sucediendo con los telefonos fisicos Thomson phones (model ST 2022) y GrandStream Budge Tone 200, este problema ocurre sin importar que opciones le coloque para el tipo de nateo dentro de los telefonos, Incluso he utilizado stun con stunserver.org o con el servidor de stun de ekiga, los telefonos se registran y pueden hacer y recibir llamadas, pero no hay audio cuando atiendes la llamada.
>
> Con kamctl ul show, puedes ver que han registrado el Contact con su IP local y el Received con la IP publica y los puertos para el NAT
> La unica diferencia con los Linksys que si funcionan es que los Linksys registran el Contact con la IP publica.
>
> Aqui se puede ver dos telefonos con NAT en el proxy
>
> Domain:: location table=512 records=2 max_slot=1
> AOR:: 20000004 en 212.4.107.250
> Contact:: sip:20000004 en 192.168.254.110:5060;transport=udp;user=phone Q=
> Expires:: 1150
> Callid:: 72ed03f6d2f390f9 en 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 en 212.4.107.250
> Contact:: sip:20000000 en 192.168.254.101:5060;user=phone Q=
> Expires:: 2945
> Callid:: 17fe-c0a80101-5-1 en 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
>
>
> Estoy utilizando rtpproxy y no hay ningun error en el log que indique que el rtpproxy no esta funcionando, de hecho haciendo un SIP trace muestra al rtpproxy seteando puertos para el audio.
> Ejecuto el rtpproxy con este comando:
>
> rtpproxy -l 212.4.107.250 -s udp:localhost:7722 -F
>
> Cualquier ayuda sera apreciada, llevo dos semanas buscando una solucion
>
> Adjunto mi kamailio.cfg para que puedan mirarlo, al final de este mensaje voy a adjuntar el SIP Trace de una llamada entre dos telefonos detras de una NAT (un Thomson y un GrandStream) en caso que puedan ayudarme a decifrar que esta mal aqui:
>
> este es mi cfg
> **************************************************************************************************
>
> #
> # $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 en 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 en 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();
> ##}
> }
>
> **************************************************************************************************
> **************************************************************************************************
>
> Aqui va el SIP Trace para una llamada de telefonos fisicos NATed to NATed:
> **************************************************************************************************
> U +0.161561 212.4.97.115:35379 -> 212.4.107.250:5060
> INVITE sip:20000000 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>
> Contact:<sip:20000004 en 192.168.254.110:5060;transport=udp;user=phone>
> Supported: replaces, timer, path
> Call-ID: c177cae013da224d en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>
> Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>
> Contact:<sip:20000004 en 212.4.97.115:35379;transport=udp;user=phone>
> Supported: replaces, timer, path
> Call-ID: c177cae013da224d en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>
> Call-ID: c177cae013da224d en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
> Call-ID: c177cae013da224d en 192.168.254.110
> CSeq: 29653 INVITE
> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
> Contact:<sip:20000000 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
> Call-ID: c177cae013da224d en 192.168.254.110
> CSeq: 29653 INVITE
> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
> Contact:<sip:20000000 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
> Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
> Call-ID: c177cae013da224d en 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 en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
> Contact:<sip:20000004 en 192.168.254.110:5060;transport=udp;user=phone>
> Supported: path
> Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
> Contact:<sip:20000004 en 212.4.97.115:35379;transport=udp;user=phone>
> Supported: path
> Call-ID: c177cae013da224d en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
> Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
> Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
> Call-ID: c177cae013da224d en 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 en 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 en 212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e
> To:<sip:20000000 en 212.4.107.250;user=phone>;tag=c0a80101-21188
> Call-ID: c177cae013da224d en 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 en 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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>
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