Harry, Great that you like the document! As with all other open source efforts, we do this in our spare time and with no financial rewards... You are very welcome to submit ser.cfg snippets, your own documentation, or even a full text describing the things you miss in the ONsip.org documentation. That's the way to do the "fixings." The wish-list of wanted stuff is quite long already, so just pointing out the gaping holes is not enough, I'm afraid ;-) The contact info for the document: gettingstarted@onsip.org g-)
harry gaillac wrote:
hello,
rtpproxy is not documented unless you look at sources
it's a good idea but onsip.org need some fixings too many people waste time to ask for the same problems on mailing list.
documentation is not updated or missing. you can reply me: "do you want to write how-to ?"
onsip.org help me however i spent time (not waste time) to understand why i could not call others domains or get rtp streams with proxy rtp.
Regards
Harry
--- "Greger V. Teigre" greger@teigre.com a écrit :
I have registered the following issue with a suggestion for README text: http://bugs.sip-router.org/browse/SER-45
I encourage everybody to register documentation needs!! g-)
harry gaillac wrote:
look at rtpp_defines.h re-compile rtpproxy. put a rtpproxy script in /etc/rc.d/init.d redhat you must add option in this script if ser run on
the
same box. So look at your ser.cfg script if you want call
others
domains or your own domain
Harry
--- Freeman hykh080@yahoo.com.hk a écrit :
How can i get more information about config port range of rtpproxy, because not found in onsip.org !
Tks a Lot ! Freeman
harry gaillac gaillacharry@yahoo.fr »¡¡GHello,
If ser and rtpproxy run on the same box your
ser.cfg
is wrong.
you need to define rtp range ports when you
compile
rtpproxy according to the rules of you fiwerall
Look at onsip.org for help !
Harry --- Freeman a écrit :
Hi,
I installed "Ser-0.8.14" + "nathelper" + "rtpproxy" in RedHat 9.0, and add "./rtpproxy"
in
startup script, when i tried both sipsoftphone
in
the internet behind NAT that can connect but "no voice" and auto hang-up, but when both
sipsoftphone
in local network that no problem !
Which port range i need release for rtpproxy in
the
firewall ?
my ser.cfg config as below :
# ----------- global configuration parameters
#debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no */ #log_stderror=yes #debug=4 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"
# ------------------ module loading
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/nathelper.so"
# Uncomment this if you want digest
authentication
# mysql.so must be loaded ! #loadmodule
"/usr/local/lib/ser/modules/dbtext.so"
#loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule
"/usr/local/lib/ser/modules/auth_db.so"
#loadmodule
"/usr/local/lib/ser/modules/mysql.so"
# ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) #modparam("auth_db", "db_url", "db:/var/dbtext") #modparam("auth_db", "user_column", "user") #modparam("auth_db", "domain_column", "domain") #modparam("auth_db", "password_column",
"password")
#modparam("auth_db", "calculate_ha1", 1) #modparam("auth_db", "password_column_2",
"ha1_2")
modparam("registrar", "nat_flag", 6) 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") # -- 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) # ------------------------- request routing
logic
# main routing logic #define NAT_UAC_TEST_C_1918 0x01 /*
- test for occurences of RFC1918 addresses in
Contact
- header field
*/ #define NAT_UAC_TEST_RCVD 0x02 /*
- test if source address of signaling is
different
from
- address advertised in Via
*/ #define NAT_UAC_TEST_V_1918 0x04 /*
- test for occurences of RFC1918 addresses in
SDP
body */ #define NAT_UAC_TEST_S_1918 0x08 /*
- test for occurences of RFC1918 addresses top
Via
*/ 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")) { append_hf("Alex-hint: NAThelper\r\n"); fix_nated_contact(); # Rewrite contact with
source
IP of signalling if (method == "REGISTER" || ! search("^Record-Route:")) { #if (www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "1"); # }; if (method == "INVITE") { append_hf("Alex-hint: SDP rewritten\r\n"); fix_nated_sdp("3"); # Add direction=active to
SDP
}; log("LOG: Someone trying to register from
private
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