hi
I've been told to setup a 48-PRI SIP/PSTN gateway system, and want to do this as follows:
1 SER server proxying all incoming SIP connections 12 asterisk servers with 4 E1 PRIs each doing the actual SIP/PSTN gatewaying 1 database server for number routing, call details and authentication info
Questions: - How can I setup SER to forward up to 120 calls per asterisk node before switching to another one? does SER keep a list of open RTP connections to the box forwarded to? - Can multiple SER boxes be used for failover or load balancing?
thanks
roy
Hi all again. Step by step with rtpproxy.
Now everything seems to work with ser 0.8.14, I mean, UA gets registered with SER and when they try to speak the start sending RTP traffic to RTPPROXY.
The weird thing is that the RTPPROXY does not forward those packets. I mean, both clients send rtp traffic to RTPPROXY ports but the RTPPROXY does not send these packets to the destiny which is the other UA.
I sniff the server traffic and RTPPROXY receives traffic from both clients ( both with private IP ) but do not send any single packet.
Does anyone know why this could be happening ?
In /var/log/messages I find three errors:
ERROR: extract_body message body has lenght zero ERROR: force_rtp_proxy2: can't extract body from message ERROR: on_reply processing failed
But it has no sense because both UAs know who they have to talk to because they are sending media traffic to the rtpproxy instead of sending it directly to the other UA. But once the RTPPROXY receives the packets it seems that he does not know what to do with them....
As I said before I'm using SER 0.8.14 from CVS and last RTPROXY from CVS too.
My ser.cfg is like:
# # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $ # # 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 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes
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 ----------------------------------
# Uncomment this if you want to use SQL database #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"
# Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# 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 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
# ------------------------- 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; };
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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "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] { # 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(); }; }
Thanks in advanced:
Kiko
On 26-07 12:54, Roy Sigurd Karlsbakk wrote:
hi
I've been told to setup a 48-PRI SIP/PSTN gateway system, and want to do this as follows:
1 SER server proxying all incoming SIP connections 12 asterisk servers with 4 E1 PRIs each doing the actual SIP/PSTN gatewaying 1 database server for number routing, call details and authentication info
Questions:
- How can I setup SER to forward up to 120 calls per asterisk node
before switching to another one? does SER keep a list of open RTP connections to the box forwarded to?
No, SER does not keep the list of active connections. You can probably distribute calls to PSTN gateways in a round-robin fashion, that could give you similar results.
- Can multiple SER boxes be used for failover or load balancing?
Yes, but it depends on the clients used.
Jan.
Hi there,
Questions:
- How can I setup SER to forward up to 120 calls per asterisk node
before switching to another one? does SER keep a list of open RTP connections to the box forwarded to?
No, SER does not keep the list of active connections. You can probably distribute calls to PSTN gateways in a round-robin fashion, that could give you similar results.
Alternatively, maybe you can use failure routes to handle this, if your PSTN-gateway can supply meaningful responses for you to handle.
Cheers,