Are the 2 sipp instances behind nat? If so, you need to set the appropiate switch on sipp, i believe it is -bind_local the actual local ip -i the public ip
Take a look at sipp's documentation Hope it helps David On Wed, Jun 6, 2018, 10:33 Jan Skórczyński <[email protected]> wrote: > Hello, > > I'm a beginner in VOIP world so maybe this question have obvious answer... > I would like to perform series of stress tests of my Opensips server. > > Opensips instance is located on VPS at adres XXX.XXX.XXX.XXX (public IP) > and it is listening on 5060 port. I'm using custom client apps, everything > works fine, however I have some issue when I try to test with Sipp. > > I'm setting UAS on my VPS, with the following command: > > sipp -sn uas -rsa XXX.XXX.XXX.XXX:5060 -i XXX.XXX.XXX.XXX > > and UAC on my PC: > > sipp -sn uac XXX.XXX.XXX.XXX:5061 -i 192.168.0.103 > > I can see on UAS that it receives messages from my UAC, but it cannot > reply to them. > > ------------------------------ Scenario Screen -------- [1-9]: Change > Screen -- > Port Total-time Total-calls Transport > 5061 19.07 s 30 UDP > > 0 new calls during 1.004 s period 1 ms scheduler resolution > 0 calls Peak was 6 calls, after 6 s > 0 Running, 31 Paused, 4 Woken up > 90 dead call msg (discarded) > 4 open sockets > > Messages Retrans Timeout > Unexpected-Msg > ----------> INVITE 30 0 0 0 > > <---------- 180 30 0 > <---------- 200 30 14 0 > ----------> ACK E-RTD1 0 0 0 30 > > ----------> BYE 0 0 0 0 > <---------- 200 0 0 > [ 4000ms] Pause 0 0 > ------------------------------ Sipp Server Mode > ------------------------------- > > > Logs from UAC: > > ------------------------------ Scenario Screen -------- [1-9]: Change > Screen -- > Call-rate(length) Port Total-time Total-calls Remote-host > 10.0(0 ms)/1.000s 5060 73.29 s 90 > XXX.XXX.XXX.XXX:5061(UDP) > > 0 new calls during 1.004 s period 1 ms scheduler resolution > 30 calls (limit 30) Peak was 30 calls, after 3 s > 0 Running, 62 Paused, 24 Woken up > 0 dead call msg (discarded) 0 out-of-call msg (discarded) > > 3 open sockets > > Messages Retrans Timeout > Unexpected-Msg > INVITE ----------> 90 415 60 > 100 <---------- 0 0 0 0 > 180 <---------- 0 0 0 0 > 183 <---------- 0 0 0 0 > 200 <---------- E-RTD1 0 0 0 0 > ACK ----------> 0 0 > Pause [ 0ms] 0 0 > BYE ----------> 0 0 0 > 200 <---------- 0 0 0 0 > > ------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause traffic > ----- > > And here is my opensips.cfg routing script: > > route{ > xlog("Begin route"); > if (!mf_process_maxfwd_header("10")) { > xlog("Too many hops"); > send_reply("483","Too Many Hops"); > exit; > } > > if (has_totag() && ($fU!="sipp")) { > xlog("Has totag"); > # handle hop-by-hop ACK (no routing required) > if ( is_method("ACK") && t_check_trans() ) { > xlog("Method: ACK && t_check_trans. Relaying..."); > t_relay(); > exit; > } > > # sequential request within a dialog should > # take the path determined by record-routing > if ( !loose_route() ) { > # we do record-routing for all our traffic, so we should not > # receive any sequential requests without Route hdr. > send_reply("404","Not here"); > exit; > } > > if (is_method("BYE")) { > # do accounting even if the transaction fails > #do_accounting("log","failed"); > } > > # route it out to whatever destination was set by loose_route() > # in $du (destination URI). > route(relay); > exit; > } > > # CANCEL processing > if (is_method("CANCEL")) { > if (t_check_trans()) > t_relay(); > exit; > } > > # absorb retransmissions, but do not create transaction > t_check_trans(); > > if ( !(is_method("REGISTER") ) ) { > if (is_myself("$fd")) { > } else { > # if caller is not local, then called number must be local > if (!is_myself("$rd")) { > send_reply("403","Relay Forbidden"); > exit; > } > } > } > > # preloaded route checking > if (loose_route()) { > xlog("L_ERR", > "Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]"); > if (!is_method("ACK")) > send_reply("403","Preload Route denied"); > exit; > } > > # record routing > if (!is_method("REGISTER|MESSAGE")) > record_route(); > > # account only INVITEs > if (is_method("INVITE")) { > create_dialog(); > engage_media_proxy(); > #do_accounting("log"); > } > if (!is_myself("$rd")) { > append_hf("P-hint: outbound\r\n"); > # if you have some interdomain connections via TLS > ## CUSTOMIZE IF NEEDED > ##if ($rd=="tls_domain1.net" > ## || $rd=="tls_domain2.net" > ##) { > ## force_send_socket(tls:127.0.0.1:5061); # CUSTOMIZE > ##} > route(relay); > } > > # requests for my domain > if (is_method("PUBLISH|SUBSCRIBE")) { > send_reply("503", "Service Unavailable"); > exit; > } > > if (is_method("REGISTER")) { > xlog("NEW REGISTER!"); > if ($proto == "tls") > setflag(TCP_PERSISTENT); > if (!save("location")) > sl_reply_error(); > > exit; > } > > if ($rU==NULL) { > # request with no Username in RURI > send_reply("484","Address Incomplete"); > exit; > } > > # do lookup with method filtering > if (!lookup("location","m")) { > xlog("not found"); > t_reply("404", "Not Found"); > exit; > } else { > xlog("FOUND"); > } > > # when routing via usrloc, log the missed calls also > #do_accounting("log","missed"); > route(relay); > } > > > route[relay] { > # for INVITEs enable some additional helper routes > if (is_method("INVITE")) { > t_on_branch("per_branch_ops"); > t_on_reply("handle_nat"); > t_on_failure("missed_call"); > } > > if (!t_relay()) { > send_reply("500","Internal Error"); > } > exit; > } > > branch_route[per_branch_ops] { > xlog("new branch at $ru\n"); > } > > onreply_route[handle_nat] { > xlog("incoming reply\n"); > } > > > failure_route[missed_call] { > if (t_was_cancelled()) { > exit; > } > > # uncomment the following lines if you want to block client > # redirect based on 3xx replies. > ##if (t_check_status("3[0-9][0-9]")) { > ##t_reply("404","Not found"); > ## exit; > ##} > } > > > What I'm doing wrong? How can I test my Opensips instance? I would like to > perform not only sip stress test but also test with some media transport. > Should I run UAC from public IP too? > > Thanks in advance for reply. > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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