This thesis is concentrated on the signalling aspect of multimedia session establishment, and more precisely on its impact to the IP telephony service quality. Loosely speaking processing of signalling information in servers on signalling path have influence on reliability and call setup time. So question is how to make IP telephony more reliable, scalable and in the same time easy to maintain. Solution is definitely introduction of parallel servers in signaling path. This approach obviously increase reliability and throughput. This thesis propose direct routing approach to SIP proxy load balancing. Direct routing in combination with failover techniques, also analyzed in this thesis, is probably the best choice for introduction of a redundant SIP proxy servers. Approach is based on a standard IPVS (IP Virtual Server) modules distributed within GNU/Linux kernel. IPVS is additionally extended with SIP scheduler especially designed for purpose of SIP traffic distribution across server farm. Failover solution is based on VRRP (Virtual Router Redundancy Protocol), and it is not in depth analyzed in this thesis. Proof of quality of direct routing concept is presented through series of measurement results. Measurements are conducted on a test network and by the principals of the Sip Stone proposal. The measurement result show that system capacity is proportional to the number and capacity of each server added to the SIP proxy farm.