[OpenSIPS-Users] Opensips 488 Not acceptable here
Bogdan-Andrei Iancu
bogdan at opensips.org
Thu Jun 6 19:47:07 CEST 2013
Hello Bogdan,
As your script does not generate the 488, it is for sure you get it from
the endpoints. As you already figured out, usually a 488 means a codec
in-compatibility between the 2 end points.
As this is completely unrelated to the proxy (not even the usage of
rtpproxy may break the codec stuff), and as it is completely random, I
would suggest taking this issue to the pjsua guys - in what conditions
the client fires the 488 - it is only based on the codecs ?
BTW, are you sure that in all your case, the 2 end points do advertise
the same list of codecs (like in all the calls ?)
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06/06/2013 08:26 PM, Bogdan Chifor wrote:
> Hello,
>
> I am using OpenSIPs as a SIP server solution.I just finished a server
> configuration which permits NAT traversal (I used RTP proxy)(on Debian).
> As a SIP client application I am using CSipSimple(Android) (which uses
> PJSUA as backend solution).
> The setup works fine and the NAT traversal problem is solved. (I am
> able to call from a 3G network provider to my wireless LAN-it uses NAT
> on both sides and it works great).
> However sometimes I am getting the 488 Not acceptable here error.
> Sometimes this error appears even when both of the phones are in the
> same LAN.
>
> Also this error appeared when I called from the 3G network operator to
> my wireless LAN.(when I called from my wireless LAN to the 3G operator
> it worked fine).
>
> The conclusion is that I cannot replicate this error every time.
>
> I know that is a SDP issue.
>
> Here is my opensips.cfg
>
> #
> # $Id: nathelper.cfg 9345 2012-10-18 20:24:22Z osas $
> #
> # simple quick-start config script including nathelper support
>
> # This default script includes nathelper support. To make it work
> # you will also have to install Maxim's RTP proxy. The proxy is enforced
> # if one of the parties is behind a NAT.
> #
> # If you have an endpoing in the public internet which is known to
> # support symmetric RTP (Cisco PSTN gateway or voicemail, for example),
> # then you don't have to force RTP proxy. If you don't want to enforce
> # RTP proxy for some destinations than simply use t_relay() instead of
> # route(1)
> #
> # Sections marked with !! Nathelper contain modifications for nathelper
> #
> # NOTE !! This config is EXPERIMENTAL !
> #
> # ----------- global configuration parameters ------------------------
>
> debug=5
> log_stderror=no
> log_facility=LOG_LOCAL0
> #log_name=opensips.log
>
> fork=yes
> children=4
>
> /* uncomment the following lines to enable debugging */
> #debug=6
> #fork=no
> #log_stderror=yes
>
>
> check_via=no # (cmd. line: -v)
> dns=no # (cmd. line: -r)
> rev_dns=no # (cmd. line: -R)
>
> #port=5060
> listen=udp:195.95.167.214:8899 <http://195.95.167.214:8899>
> #advertised_address="195.95.167.194"
> #advertised_port=5060
> port=8899
>
>
> children=4
>
> disable_tcp=yes
>
> alias=voip.certsign.ro <http://voip.certsign.ro>
> alias=195.95.167.214
> #alias=192.168.185.26
> # ------------------ module loading ----------------------------------
>
> #set module path
> mpath="/usr/lib/opensips/modules/"
>
> # Uncomment this if you want to use SQL database
> loadmodule "db_mysql.so"
>
> loadmodule "sl.so"
> loadmodule "tm.so"
> loadmodule "signaling.so"
> loadmodule "rr.so"
> loadmodule "maxfwd.so"
> loadmodule "usrloc.so"
> loadmodule "registrar.so"
> loadmodule "textops.so"
> loadmodule "mi_fifo.so"
> loadmodule "sipmsgops.so"
> loadmodule "dialog.so"
> loadmodule "avpops.so"
> loadmodule "domain.so"
> #loadmodule "xlog.so"
> loadmodule "acc.so"
>
> # Uncomment this if you want digest authentication
> # db_mysql.so must be loaded !
> loadmodule "auth.so"
> loadmodule "auth_db.so"
>
> # !! Nathelper
> loadmodule "nathelper.so"
> loadmodule "rtpproxy.so"
>
> # ----------------- setting module-specific parameters ---------------
>
> # -- mi_fifo params --
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
>
> # -- usrloc params --
> modparam("usrloc", "db_mode", 2)
> modparam("usrloc", "db_url", "mysql://opensips:qwe123@localhost/opensips")
>
>
> # -- 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")
> modparam("auth_db", "db_url",
> "mysql://opensips:qwe123@localhost/opensips")
> modparam("auth_db", "load_credentials", "")
>
> db_default_url="mysql://opensips:qwe123@localhost/opensips"
>
>
> #
> # !! Nathelper
> #
> modparam("usrloc", "nat_bflag", 6)
> modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
> modparam("nathelper", "sipping_bflag", 8)
> modparam("nathelper", "received_avp", "$avp(i:801)")
>
> # RTPProxy setup
> #modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7789
> <http://127.0.0.1:7789>")
> modparam("nathelper", "force_socket", "udp:127.0.0.1:7789
> <http://127.0.0.1:7789>")
>
> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7789
> <http://127.0.0.1:7789>")
> modparam("rtpproxy", "rtpproxy_autobridge", 1)
> modparam("rtpproxy", "rtpproxy_timeout", "0.5")
> modparam("rtpproxy", "rtpproxy_retr", 3)
>
>
> #
> # ------- dialog --------
> #**
> modparam("dialog", "db_mode", 1)
> modparam("dialog", "db_update_period", 30)
> #modparam("dialog", "dlg_flag", 4)
> modparam("dialog", "dlg_match_mode", 1)
>
> #
> # --> avpops params -----
> #**
> modparam("avpops", "avp_table", "usr_preferences")
> modparam("avpops", "use_domain", 1)
>
>
>
> # ************
> # ----- presence params -----
> /* uncomment the following lines if you want to enable presence */
> #modparam("presencepresence_xml", "db_url",
> # "mysql://opensips:opensipsrw@localhost/opensips")
> #modparam("presence_xml", "force_active", 1)
> #modparam("presence", "server_address", "sip:localhost:5060")
>
>
> # ------------------------- 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");
> exit;
> };
> if (msg:len >= 2048 ) {
> sl_send_reply("513", "Message too big");
> exit;
> };
>
> # !! Nathelper
> # Special handling for NATed clients; first, NAT test is
> # executed: it looks for via!=received and RFC1918 addresses
> # in Contact (may fail if line-folding is used); also,
> # the received test should, if completed, should check all
> # vias for rpesence of received
> #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 (is_method("REGISTER") ||
> !is_present_hf("Record-Route")) {
> log("LOG:Someone trying to register from
> private IP, rewriting\n");
> # This will work only for user agents that
> support symmetric
> # communication. We tested quite many of them
> and majority is
> # smart enough to be symmetric. In some phones
> it takes a
> # configuration option. With Cisco 7960, it is
> called
> # NAT_Enable=Yes, with kphone it is called
> "symmetric media" and
> # "symmetric signalling".
>
> # Rewrite contact with source IP of signalling
> fix_nated_contact();
> if ( is_method("INVITE") ) {
> log("@DEBUG:FIX SDP");
> fix_nated_sdp("2"); # Add
> direction=active to SDP
> };
> force_rport(); # Add rport parameter to
> topmost Via
> setbflag(6); # Mark as NATed
>
> # if you want sip nat pinging
> # setbflag(8);
> };
> #};
>
> # 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);
> exit;
> };
> # 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 (!is_method("REGISTER"))
> record_route();
>
> if (!uri==myself) {
> # mark routing logic in request
> append_hf("P-hint: outbound\r\n");
> route(1);
> exit;
> };
>
> # 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 (is_method("REGISTER")) {
>
> # Uncomment this if you want to use digest
> authentication
> #if (!www_authorize("siphub.org
> <http://siphub.org>", "subscriber")) {
> # www_challenge("siphub.org
> <http://siphub.org>", "0");
> # return;
> #};
>
> save("location");
> exit;
> };
>
> lookup("aliases");
> if (!uri==myself) {
> append_hf("P-hint: outbound alias\r\n");
> route(1);
> exit;
> };
>
> # native SIP destinations are handled using our USRLOC DB
> if (!lookup("location")) {
> sl_send_reply("404", "Not Found");
> exit;
> };
> };
> append_hf("P-hint: usrloc applied\r\n");
> log("@DEBUG:LINE BEFORE ROUTE[1]");
> 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");
> exit;
> };
>
> # if client or server know to be behind a NAT, enable relay
> if (isbflagset(6)){
>
> if (is_method("INVITE")) {
> log("@DEBUG:INVITE METHOD");
> if (has_body("application/sdp")) {
> log("@DEBUG:application/sdp");
> if (rtpproxy_offer("195.95.167.214")) {
> t_on_reply("2");
> log("@DEBUG:RTP PROXY OFFER");
> }
> } else {
> t_on_reply("3");
> }
> }
> if (is_method("ACK") && has_body("application/sdp"))
> rtpproxy_answer();
> };
>
> # 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();
> };
> }
>
> # !! Nathelper
> onreply_route[1] {
> # NATed transaction ?
> if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
> fix_nated_contact();
> rtpproxy_answer();
> # 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();
> };
> }
> onreply_route[2]
> {
>
> if (has_body("application/sdp"))
> rtpproxy_answer("195.95.167.214");
> }
>
> onreply_route[3]
> {
>
> if (has_body("application/sdp"))
> rtpproxy_offer("195.95.167.214");
> }
>
>
>
> Please help me.
>
> Thank you very much,
>
> Chifor Bogdan
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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