An improved B2BUAWM approach for VoIP Infrastructure Fernando Barreto Coordenation and Management of Information Technology Federal University of Technology – Parana (UTFPR) Apucarana, Brazil fbarreto@utfpr.edu.br Abstract—This article presents a study for VoIP technology to elaborate a differentiated VoIP infrastructure with the proposed approach ProxySIP Bridge. This approach enhances the Back-to- Back User Agent With Media (B2BUAWM) approach in order to isolate the voice traffic from the remaining general purpose traffic and to enable direct voice traffic whenever possible. A differentiated VoIP infrastructure with ProxySIP Bridge is pre- sented, which is an alternative to B2BUAWM original approach. Keywords-component; VoIP infrastructure, B2BUAWM, ProxySIP Bridge I. INTRODUCTION With the advances of network computer technologies, the IP network infrastructure can provide a voice over IP (VoIP) service at an acceptable level of quality. The VoIP service can significantly reduce the phone call cost when compared to the standard public switch telephony network. This technology is helpful to companies composed of many affiliates, who recognize it as a chance to significantly reduce telephone costs. In the academic environment, this reduction is also interesting, mainly if a university is composed of various campuses. There are many important points that should be considered in a VoIP infrastructure such as Firewall, Network Address Translation (NAT) and Quality of Service (QoS). Generally, these points make the VoIP infrastructure very difficult to implement. In order to achieve a facilitated VoIP infrastructure, the B2BUAWM approach can solve Firewall and NAT problems, because it needs only small modifications on Firewall/NAT and no support from VoIP software client. However, B2BUAWM presents some problems of overhead which can be solved with an enhanced B2BUAWM approach named ProxySIP Bridge. The ProxySIP Bridge allows a differentiated VoIP infrastructure which deviates the voice traffic from Firewall/NAT, reduces the number of hops needed to forward the voice traffic between the end users equipment and, whenever possible, can enable voice traffic to occur directly between them. The ProxySIP Bridge approach was implemented using open source software, which enables its adaptation by the academic community. II. VOICE OVER IP The establishment of a VoIP call needs a signaling protocol. This signaling is made by Session Initiation Protocol (SIP) [4]. In order to transport the data stream containing the digital audio voice the SIP use the Real-Time Transport Protocol (RTP) [6] and, to monitor the RTP traffic, they use the RTP Control Protocol (RTCP) [6]. A. SIP The SIP protocol is specified in [4]. A hardware equipment that acts on behalf of a user and implements the SIP protocol is denominated User Agent (UA). The main methods defined to establish and finish a VoIP call are: INVITE, ACK and BYE. The INVITE and BYE methods need an answer from remote UA to confirm the method operation. The INVITE method carries SIP information to identify the source UA, the destination UA and the Session Description Protocol (SDP) [5], which carries additional information from source UA, like Codec details, network IP address and port number for RTP session. When the destination UA accepts the INVITE message, it sends a message with an OK code containing a similar SDP with information for RTP session. When the source UA receives the OK message, it sends an ACK message to the destination UA to confirm the RTP session and begins the voice transmission. When the destination UA receives the ACK, it also begins the voice transmission. Further details on SIP message flows and methods are presented in [4]. In a VoIP infrastructure with many UAs that belong to the same SIP domain, a centralized Proxy service [4] is recommended for SIP domain, here called ProxySIP. The objective is to facilitate the UA localization inside a SIP domain. The ProxySIP controls, authenticates the UAs and is also responsible for routing the SIP messages. This enables the management of SIP messages from INVITE to BYE (SIP transactions and SIP dialog) [4]. B. Codec and Quality of Service Concepts The voice RTP session is configured according to the coder/decoder (Codec) information exchanged through SDP. More details about Codecs can be found in [14]. The quality of a Codec can be measured by the Mean Opinion Score (MOS) index. Table I presents a few standard Codecs. The Voice Bandwidth is the constant bit rate needed for Codec operation. The Total Bandwidth takes into consideration the overhead of all TCP/IP headers in an Ethernet network (78 bytes) and also a 978-1-4577-1792-5/11/$26.00 ©2011 IEEE