Pushed by the growth of the Internet, the IP Telephony conquered a great attention of the market and big suppliers, with promises of cost reductions on operation, management, provisioning, maintenance and billing. Different protocols were developed for providing VoIP such as H.323, MGCP, Megaco and SIP. SIP has been highlighted for being a text based protocol, extensible, independent of the transport protocol, therefore more flexible and simpler than your competitor, the H.323. SIP (Session Initiation Protocol) is a signaling protocol used for establish, modify and terminate sessions. It can be used for voice calls over IP (VoIP) or to exchange instant messaging, among other applications. It has been developed originally in 1996 and has been standardized by IETF in 1999. In this work, the performance of SIP protocol for establishing VoIP calls will be estimated, since there are a lot of papers focalizing in the voice quality and few treated the signaling [3]. Experimental environments will be used for varying parameters like delay, packet loss, jitter, bandwidth and transport protocol, allowing to verify how there parameters affect separately the post-dial delay, post-pickup delay and call release delay.