Hello All:

 

You will find enclosed our ser.cfg file that I have adapted from a very generous SIP user from the Internet.
  
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.

 

We have checked the etherreal output while trying to make a call and we are having this type of errors on the etherreal:

 

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debug=3
fork=yes
log_stderror=no

check_via=no
dns=no
rev_dns=no

listen=XXX.XXX.XXX.XXX           # INSERT YOUR IP ADDRESS HERE
# port=5060
children=3

alias="HOST.DOMAINNAME.COM"
alias="XXX.XXX.XXX.XXX"

fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:PASSWORD@localhost/ser"

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/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"

modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:PASSWORD@localhost/ser")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "password_column", "password")

modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)  
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")

modparam("usrloc", "db_mode", 2)

modparam("registrar", "nat_flag", 6)

modparam("rr", "enable_full_lr", 1)
#---Accounting params
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 1)

# -------------------------  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;
 };
        # !! 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 (method == "REGISTER" || ! search("^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".

      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;
 };

 setflag(1);
 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("HOST.DOMAINNAME.COM", "subscriber")) {
    www_challenge("HOST.DOMAINNAME.COM", "0");
    break;
   };

   save("location");
   break;
  };

  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
  if (!lookup("location")) {
   sl_send_reply("404", "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();
 };
}

# !! Nathelper
onreply_route[1] {
    # NATed transaction ?
    if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
        fix_nated_contact();
 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();
    };
}
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