<DIV>Hello All:</DIV> <DIV> </DIV> <DIV>You will find enclosed our ser.cfg file that I have adapted from a very generous SIP user from the Internet.<BR> <BR>This ser.cfg file work perfectly with Audio (IP Phone and Softphone). But when I am trying to use a Video Phone (we did try with GrandStream GXV-3000 Video Phone), it just giving us Audio but NOT Video. I did also try the same GrandStream GXV-3000 Video Phone with FreeWorldDialup and it is working fine with Video.</DIV> <DIV> </DIV> <DIV>We have checked the etherreal output while trying to make a call and we are having this type of errors on the etherreal:</DIV> <DIV> </DIV> <DIV>-----</DIV> <DIV> <DIV>RTP Protocol</DIV> <DIV>Source IP (The IP of the Video Phone) and Destination IP (The IP of the SER Server) ------­ Payload Type = Unknown (126)</DIV> <DIV> </DIV> <DIV>ICMP Protocol</DIV> <DIV>Source IP (The IP of the SER Server) and Destination IP (The
IP of the Video Phone) ------­Destination Unreachable (Port Unreacheable)</DIV> <DIV> </DIV> <DIV>These 2 errors repeated six time before it stop with a good signal of:</DIV> <DIV> </DIV> <DIV>Payload Type = ITU-T G.711 PCMU</DIV> <DIV> </DIV> <DIV>Then the audio start but without audio.<BR>---</DIV> <DIV><BR>Would you please tell us what is going wrong with this ser.cfg file? your answer is greatly appreciated.<BR> <BR>Many thanks,<BR> <BR>Steven Wong</DIV></DIV> <DIV> </DIV> <DIV>--------------------------------------------------------------------------</DIV> <DIV>debug=3<BR>fork=yes<BR>log_stderror=no</DIV> <DIV>check_via=no<BR>dns=no<BR>rev_dns=no</DIV> <DIV>listen=XXX.XXX.XXX.XXX # INSERT YOUR IP ADDRESS HERE<BR># port=5060<BR>children=3</DIV> <DIV>alias="<SPAN id=lw_1172198623_0 style="BORDER-BOTTOM: #0066cc 1px
dashed">HOST.DOMAINNAME.COM</SPAN>"<BR>alias="XXX.XXX.XXX.XXX"</DIV> <DIV>fifo="/tmp/ser_fifo"<BR>fifo_db_url="mysql://ser:PASSWORD@localhost/ser"</DIV> <DIV>loadmodule "/usr/local/lib/ser/modules/mysql.so"<BR>loadmodule "/usr/local/lib/ser/modules/sl.so"<BR>loadmodule "/usr/local/lib/ser/modules/tm.so"<BR>loadmodule "/usr/local/lib/ser/modules/rr.so"<BR>loadmodule "/usr/local/lib/ser/modules/maxfwd.so"<BR>loadmodule "/usr/local/lib/ser/modules/usrloc.so"<BR>loadmodule "/usr/local/lib/ser/modules/registrar.so"<BR>loadmodule "/usr/local/lib/ser/modules/auth.so"<BR>loadmodule "/usr/local/lib/ser/modules/auth_db.so"<BR>loadmodule "/usr/local/lib/ser/modules/uri.so"<BR>loadmodule "/usr/local/lib/ser/modules/uri_db.so"<BR>loadmodule "/usr/local/lib/ser/modules/nathelper.so"<BR>loadmodule "/usr/local/lib/ser/modules/textops.so"<BR>loadmodule "/usr/local/lib/ser/modules/acc.so"</DIV> <DIV>modparam("auth_db|uri_db|usrloc", "db_url",
"mysql://ser:PASSWORD@localhost/ser")<BR>modparam("auth_db", "calculate_ha1", 1)<BR>modparam("auth_db", "password_column", "password")</DIV> <DIV>modparam("nathelper", "natping_interval", 30) <BR>modparam("nathelper", "ping_nated_only", 1) <BR>modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")</DIV> <DIV>modparam("usrloc", "db_mode", 2)</DIV> <DIV>modparam("registrar", "nat_flag", 6)</DIV> <DIV>modparam("rr", "enable_full_lr", 1)<BR>#---Accounting params<BR>modparam("acc", "log_level", 1)<BR>modparam("acc", "log_flag", 1)</DIV> <DIV># ------------------------- request routing logic -------------------</DIV> <DIV># main routing logic</DIV> <DIV>route{</DIV> <DIV> # initial sanity checks -- messages with<BR> # max_forwards==0, or excessively long requests<BR> if (!mf_process_maxfwd_header("10")) {<BR> sl_send_reply("483","Too Many Hops");<BR> break;<BR> };<BR> if (msg:len >=
max_len ) {<BR> sl_send_reply("513", "Message too big");<BR> break;<BR> };<BR> # !! Nathelper<BR> # Special handling for NATed clients; first, NAT test is<BR> # executed: it looks for via!=received and RFC1918 addresses<BR> # in Contact (may fail if line-folding is used); also,<BR> # the received test should, if completed, should check all<BR> # vias for rpesence of received<BR> if (nat_uac_test("3")) {<BR> # Allow RR-ed requests, as these may indicate that<BR> # a NAT-enabled proxy takes care of it; unless it is<BR> # a REGISTER<BR> if (method == "REGISTER" || ! search("^Record-Route:")) {<BR> log("LOG: Someone trying to register from private IP, rewriting\n");</DIV> <DIV> # This will work only for user agents that support symmetric<BR> #
communication. We tested quite many of them and majority is<BR> # smart enough to be symmetric. In some phones it takes a configuration<BR> # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is<BR> # called "symmetric media" and "symmetric signalling".</DIV> <DIV> fix_nated_contact(); # Rewrite contact with source IP of signalling<BR> if (method == "INVITE") {<BR> fix_nated_sdp("1"); # Add direction=active to SDP<BR> };<BR> force_rport(); # Add rport parameter to topmost Via<BR> setflag(6); # Mark as NATed<BR> };<BR> };</DIV> <DIV> # we record-route all messages -- to make sure that<BR> # subsequent messages will go through our proxy; that's<BR> #
particularly good if upstream and downstream entities<BR> # use different transport protocol<BR> if (!method=="REGISTER") record_route(); </DIV> <DIV> # subsequent messages withing a dialog should take the<BR> # path determined by record-routing<BR> if (loose_route()) {<BR> # mark routing logic in request<BR> append_hf("P-hint: rr-enforced\r\n"); <BR> route(1);<BR> break;<BR> };</DIV> <DIV> setflag(1);<BR> if (!uri==myself) {<BR> # mark routing logic in request<BR> append_hf("P-hint: outbound\r\n"); <BR> route(1);<BR> break;<BR> };</DIV> <DIV> # if the request is for other domain use UsrLoc<BR> # (in case, it does not work, use the following command<BR> # with proper names and addresses in it)<BR> if (uri==myself) {</DIV> <DIV> if (method=="REGISTER") {</DIV> <DIV># Uncomment this if you want to use
digest authentication<BR> if (!www_authorize("<SPAN id=lw_1172198623_1 style="BORDER-BOTTOM: #0066cc 1px dashed">HOST.DOMAINNAME.COM</SPAN>", "subscriber")) {<BR> www_challenge("<SPAN id=lw_1172198623_2 style="BORDER-BOTTOM: #0066cc 1px dashed">HOST.DOMAINNAME.COM</SPAN>", "0");<BR> break;<BR> };</DIV> <DIV> save("location");<BR> break;<BR> };</DIV> <DIV> lookup("aliases");<BR> if (!uri==myself) {<BR> append_hf("P-hint: outbound alias\r\n"); <BR> route(1);<BR> break;<BR> };</DIV> <DIV> # native SIP destinations are handled using our USRLOC DB<BR> if (!lookup("location")) {<BR> sl_send_reply("404", "Not Found");<BR> break;<BR> };<BR> };<BR> append_hf("P-hint: usrloc applied\r\n");
<BR> route(1);<BR>}</DIV> <DIV>route[1] <BR>{<BR> # !! Nathelper<BR> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){<BR> sl_send_reply("479", "We don't forward to private IP addresses");<BR> break;<BR> };<BR> <BR> # if client or server know to be behind a NAT, enable relay<BR> if (isflagset(6)) {<BR> force_rtp_proxy();<BR> };</DIV> <DIV> # NAT processing of replies; apply to all transactions (for example,<BR> # re-INVITEs from public to private UA are hard to identify as<BR> # NATed at the moment of request processing); look at replies<BR> t_on_reply("1");</DIV> <DIV> # send it out now; use stateful forwarding as it works reliably<BR> # even for UDP2TCP<BR> if (!t_relay()) {<BR> sl_reply_error();<BR> };<BR>}</DIV> <DIV># !!
Nathelper<BR>onreply_route[1] {<BR> # NATed transaction ?<BR> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {<BR> fix_nated_contact();<BR> force_rtp_proxy();<BR> # otherwise, is it a transaction behind a NAT and we did not<BR> # know at time of request processing ? (RFC1918 contacts)<BR> } else if (nat_uac_test("1")) {<BR> fix_nated_contact();<BR> };<BR>}<BR>--------------------------------------------------------------------------</DIV><p> 
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