Re: [Serusers] ser + rtpproxy - call disconnected

Hi, In fact, my PSTN GW does not support STUN. If it support STUN, then I will not use rtpproxy. The following problem happens: I try to use a fix IP for PSTN GW to register with SER, if I don't enable rtpproxy, only ringing tones could be heard, but no voice passing through it. Why does this happen? Another question is that where could I download and install ngrep? If I use a capture program to capture all packets, could I replace ngrep with it? Thank you for the advice. Best Regards, Thomas ----- Original Message ----- From: Vitaly Nikolaev To: 'support' ; serusers(a)lists.iptel.org Sent: Thursday, January 03, 2002 3:37 AM Subject: RE: [Serusers] ser + rtpproxy - call disconnected Even if half of all RTP packets get lost you will just have bad quality of call. In your case it very looks like you have problem with ACK DO ngrep and see if PSTN GW receive ACK packet after OK (otherwise GW will repeat OK few times and then disconnect call) if it is an issue just do search in this list with keyword ACK ------------------------------------------------------------------------------ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of support Sent: Saturday, February 26, 2005 12:24 AM To: serusers(a)lists.iptel.org Subject: [Serusers] ser + rtpproxy - call disconnected Hi, When I try to make a call using rtpproxy and ser-0.8.14 (SIP UA <---> PSTN), most of the time the call will be disconnected within 1 minutes. Sometimes, it will be so unstable that the call be disconnected after 15 seconds. The above scenarios happens Will this problem related to unstable rtpproxy? Because I know that all packets will route through the server. once rtp packet get loss, the call will be disconnected. Is this correct? Perhaps it depends on my ser.cfg. Thomas My ser.cfg: # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen="" port=5060 children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- # 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 modparam("rr", "enable_full_lr", 1) # -- Nathelper params -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic route{ # ----------------------------------------------- # Sanity Check Section # ----------------------------------------------- # 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; }; # ----------------------------------------------- # NOTIFY Keep-Alive Section # ----------------------------------------------- if ((method=="NOTIFY") && search("^Event: keep-alive")) { sl_send_reply("200","OK"); break; }; # Nathelper if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); 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) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; # if the dialed number lies in the range 35891500-35891799, don't forward it to T1 Trunk GW if ((uri=~"^sip:(852|)358915[0-9][0-9]@") || (uri=~"^sip:(852|)358916[0-9][0-9]@") || (uri=~"^sip:(852|)358917[0-9][0-9]@")) { if (uri=~"^sip:852*") { strip(3); }; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB # Call Routing Section if (!lookup("location")) { if (uri=~"^sip:(852|)[0-9]{8}@") { # Send to PSTN Gateway route(2); break; }; sl_send_reply("404", "User Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } 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 client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); break; }; } # PSTN Call to T1 Trunk GW route[2] { rewritehostport(""); if (isflagset(6)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); break; }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # Not all 2xx messages have a content body so here we make sure # out Content-Length > 0 to avoid a parse error if (!search("^Content-Length:\0")) { force_rtp_proxy(); }; # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } ---------end of config -----------