[Serusers] Contact Header and SDP not rewritten
Greger V. Teigre
greger at teigre.com
Wed Apr 6 07:46:34 CEST 2005
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 at 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 at 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 at 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 at 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 at 84.203.148.146>;tag=2dc376dcd4655094..
>> To: <sip:2092 at 84.203.148.146>..
>> Contact: <sip:2093 at 84.203.148.14:5060>..
>> *==> Correctly changed to the public address and port of 2093
>> Supported: replaces..Call-ID: 44e1ae63c476fbff at 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 at 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 at 84.203.148.146 SIP/2.0..Via: SIP/2.0/UDP
>>> 172.16.3.31;branch
>>> =z9hG4bK1a48edc121f5bc1f..From: "2093"
>>> <sip:2093 at 84.203.148.146>;tag=2dc376
>>> dcd4655094..To: <sip:2092 at 84.203.148.146>..Contact:
>>> <sip:2093 at 172.16.3.31>.
>>> .Supported: replaces..Call-ID: 44e1ae63c476fbff at 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 at 84.203.148.146>;tag=2dc376dcd4655094..To:
>>> <sip:2092 at 84.
>>> 203.148.146>..Call-ID: 44e1ae63c476fbff at 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 at 84.203.148.146
>>> out_uri=sip:2092 at 1
>>> 57.190.74.151 via_cnt==1"....
>>>
>>> U 84.203.148.146:5060 -> 157.190.74.151:5060
>>> INVITE sip:2092 at 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 at 84.2
>>> 03.148.146>;tag=2dc376dcd4655094..To:
>>> <sip:2092 at 84.203.148.146>..Contact: <
>>> sip:2093 at 84.203.148.14:5060>..Supported: replaces..Call-ID:
>>> 44e1ae63c476fbf
>>> f at 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 at 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 at 84.2
>>> 03.148.146>;tag=2dc376dcd4655094..To:
>>> <sip:2092 at 84.203.148.146>..Contact: <
>>> sip:2093 at 84.203.148.14:5060>..Supported: replaces..Call-ID:
>>> 44e1ae63c476fbf
>>> f at 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 at 84.203.148.146>;tag=2dc3
>>> 76dcd4655094..To: <sip:2092 at 84.203.148.146>..Call-ID:
>>> 44e1ae63c476fbff at 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 at 84.203.148.146>;tag=2dc
>>> 376dcd4655094..To:
>>> <sip:2092 at 84.203.148.146>;tag=10bdf2044401a257..Call-ID:
>>> 44e1ae63c476fbff at 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
>> &g! t;
>>> 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 at lists.iptel.org
>>> http://lists.iptel.org/mailman/listinfo/serusers
>> Send instant messages to your online friends
>> http://uk.messenger.yahoo.com
> Send instant messages to your online friends
> http://uk.messenger.yahoo.com
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