[Serusers] Re: SER Call Forking with Cisco Routers as SIP endpoints
Psyber
psyber1 at gmail.com
Thu Feb 24 02:46:04 CET 2005
Anyone have any ideas on this? I'm considering setting up
rtpproxy as a temporary workaround (assuming that will somehow
mitigate the issue), but given my evironment it won't be a feasible
production solution. I'm still pretty baffled as to why parties can
talk between router1 and router3, router2 and router 3, but not
between router1 and router2. Further confusing the situation is the
fact that the call completes just fine when calling between router1
and router 2 and an RTP stream is established between the two routers,
just no sound. router1 and router2 have a clear path to each other.
On Thu, 17 Feb 2005 18:41:04 -0500, Psyber <psyber1 at gmail.com> wrote:
> I'm presently working on a SIP setup whereby there are 3 Cisco
> routers which each have analog phones connected to them via FXS ports.
> All 3 of these routers are connected via an underlying network. I
> have a machine hanging off from one of these routers running ser. For
> ease of labelling, I'll call these routers: router1, router2, and
> router3 (SIP server directly connected to this router via ethernet).
> I'm attempting to setup call forking using the UsrLoc database (this
> will eventually be SQL, but for the sake of the short-term I'm just
> storing UsrLoc in memory). The desired call forking setup looks
> something like this:
>
> router1 --> router2
> --> router 3
>
> router 2 --> router 1
> --> router 3
>
> router 3 --> router 1
> --> router 2
>
> I am able to complete calls between router1 and router3 (and
> vice-versa) and carry on a conversation, but when calling between
> router1 and router2 the call completes, but neither party can hear the
> other. Ironically, router1 and router2 are sitting right next to each
> other (though, connected via another router). However, The SIP proxy
> is directly connected to router3. Doing a 'debug voip rtp' I see RTP
> messages travel bidirectionally in a constant stream with correct IP
> addresses and ports until the call ends, but at no point during the
> conversation can either party hear the other. This would lead me to
> believe that something other than SIP was at play, but when I bypass
> the proxy (point the two routers directly at each other via the
> dial-peer) call completion works and both parties can hear each other
> (I set these up as SIP, not the default H.323). Below is my ser.cfg
> file and the output of 'serctl ul show' for the static UsrLoc entries
> that I've created. The routers are setup with simple dial-peers and a
> sip-ua.
>
> I've verified that there isn't any type of ACL or firewall to obstruct
> the conversation. Every router is able to reach each other router as
> well as the proxy server. I'm using private address space at present,
> but NAT isn't being done at any point. I've pondered trying rtp_proxy
> and forcing the bearer (RTP) traffic through the proxy, but that isn't
> a particularly good solution for my environment.
>
> Any help would be greatly appreciated. I'm hoping that it's just a
> case of broken logic in my ser.cfg. Please CC: this address in your
> reply as I'm not currently on the mailing list.
>
> Most of the configuration is derived from the sample configurations
> that I ran into.
>
> ---ser.cfg start---
>
> # ----------- global configuration parameters ------------------------
>
> debug=7 # debug level (cmd line: -dddddddddd)
> fork=yes
> log_stderror=yes # (cmd line: -E)
>
> /* Uncomment these lines to enter debugging mode
> #debug=7
> #fork=no
> #log_stderror=yes
> */
>
> check_via=no # (cmd. line: -v)
> dns=no # (cmd. line: -r)
> rev_dns=no # (cmd. line: -R)
> listen=192.168.1.2
> port=5060
> mhomed=1
> #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"
>
> # 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"
>
> # ----------------- 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)
>
> # ------------------------- request routing logic -------------------
>
> # main routing logic
> alias="ser"
>
> 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;
> };
>
> # 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=="INVITE") record_route();
>
> # loose-route processing
> if (loose_route()) {
> t_relay();
> 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;
> };
>
> # native SIP destinations are handled using our USRLOC DB
> if (!lookup("location")) {
> sl_send_reply("404", "Not Found");
> break;
> };
> };
> # forward to current uri now; use stateful forwarding; that
> # works reliably even if we forward from TCP to UDP
> if (!t_relay()) {
> sl_reply_error();
> };
>
> }
>
> ---end ser.cfg---
>
> ---start static UsrLoc entries---
>
> ser# ../../sbin/serctl ul show 222
> 200 OK
> <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231
> <sip:222@<router3 IP>:5060>;q=1.00;expires=1003718231
>
> ser# ../../sbin/serctl ul show 111
> 200 OK
> <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231
> <sip:111@<router3 IP>:5060>;q=1.00;expires=1003718231
>
> ser# ../../sbin/serctl ul show 333
> 200 OK
> <sip:111@<router1 IP>:5060>;q=1.00;expires=1003718231
> <sip:222@<router2 IP>:5060>;q=1.00;expires=1003718231
>
> ---end static UsrLoc entries---
>
> Thank you.
>
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