Arquivo da tag: opensips

opensips, voip

Routing básico no OpenSips

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Na configuração abaixo OpenSIPS não trata o tráfego de áudio/RTP, este é passado diretamente para o servidor Asterisk/A2Billing.

O código a seguir é o todo OpenSIPS.cfg, depois quero adicionar o artigo todo após testado completamente.

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route{
if (!mf_process_maxfwd_header("10") && $retcode==-1) {
sl_send_reply("483","Too Many Hops");
exit;
}
if (has_totag()) {
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method("INVITE")) {
record_route();
}
route(RELAY);
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
t_relay();
exit;
} else {
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
 
if (is_method("CANCEL")) {
if (t_check_trans())
t_relay();
exit;
} else if (!is_method("INVITE")) {
send_reply("405","Method Not Allowed");
xlog("$rm FAILED: $si / $ct / $fu\n");
exit;
}
if ($rU==NULL) {
sl_send_reply("484","Address Incomplete");
exit;
}

 

 

opensips, voip

Opensips Features

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Bom esse é um assunto que um dia espero dar mais detalhes, não é algo comumente utilizado e ou falado, eu diria que é mais coisa de especialistas, mais para operadoras que estão migrando sua telefonia velha “legado” para o novo padrão mundial o SIP, o que posso adiantar é que o opensips é ambicioso, diz que pode manter até 5000CPS ou seja chamadas por segundo e pode ter 3milhões de usuários isso são números bem interessantes e milhares de vezes maiores que o asterisk que foi concebido para outra coisa, não entrarei nesta conversa. Para quem quer saber algumas caracteristicas do opensips abaixo alguns features sobre o mesmo.

Some of the features that OpenSIPS brings:

  • robust and performant SIP (RFC3261) Registrar server, Location server, Proxy server and Redirect server
  • small footprint – the binary file is small size, functionality can be stripped/added via modules
  • plug&play module interface – ability to add new extensions, without touching the core, therefore assuring a great stability of core components
  • stateless and transactional statefull SIP Proxy processing
  • support for UDP/TCP/TLS/SCTP transport layers
  • IPv4 and IPv6
  • support for SRV and NAPTR DNS
  • SRV DNS failover
  • IP Blacklists
  • multi-homed (mhomed) and multi-domain support
  • flexible and powerful scripting language for routing logic
  • variables support in script – script variables, pseudo-variables (access to the SIP messages), AVPs (values persistent per SIP transactions)
  • management interface via FIFO file and unix sockets
  • authentication, authorization and accounting (AAA) via database (MySQL, Postgress, text files), RADIUS and DIAMETER
  • digest and IP authentication
  • Presence Agent support (many additional integration features)
  • XCAP support for Presence Agent
  • CPL – Call Processing Language (RFC3880)
  • SNMP – interface to Simple Network Management Protocol
  • management interface (for external integration) via FIFO file, XMLRPC or Datagram (UDP or unixsockets)
  • NAT traversal support for SIP and RTP traffic
  • ENUM support
  • PERL Programming Interface – embed your extensions written in Perl
  • Java SIP Servlet Application Interface – write Java SIP Servlets to extent your VoIP services and integrate with web services
  • load balancing with failover
  • least cost routing
  • support for replication – REGISTER offer new functions for replicating client information (real source and received socket).
  • logging capabilities – can log custom messages including any header or pseudo-variable and parts of SIP message structure.
  • modular architecture – plug-and-play module interface to extend the server’s functionality
  • gateway to sms (AT based)
  • multiple database backends – MySQL, PostgreSQL, Oracle, Berkeley, flat files and other database types which have unixodbc drivers
  • straightforward interconnection with PSTN gateways
  • dialog support (call monitoring, call termination from proxy side, call profiling)
  • XMPP gateway-ing ( transparent server-to-server translation)
  • impressive extension repository – over 70 modules are included in OpenSIPS repository

Scalability:

  • OpenSIPS can run on embedded systems, with limited resources – the performances can be up to hundreds of call setups per second
  • used a load balancer in stateless mode, OpenSIPS can handle over 5000 call setups per second
  • on systems with 4GB memory, OpenSIPS can serve a population over 300 000 online subscribers
  • system can easily scale by adding more OpenSIPS servers
  • OpenSIPS can be used in geographic distributed VoIP platforms
  • straightforward failover and redundancy