In the overwhelming majority of implementations, RTP is used to transmit VoIP traffic ("media"). Notable exception is IAX which carries both signaling and voice data over a single UDP stream, which results in fewer problems while traversing firewall and NAT devices.

For signaling, there are several alternative protocols:

• Session Initiation Protocol (SIP), an IETF newcomer gaining popularity
• H.323, the ITU's widely deployed and continually updated VoIP protocol carrying billions of minutes of traffic each month
• Skinny Client Control Protocol, proprietary protocol from Cisco
• Megaco (a.k.a. H.248) and MGCP, both media gateway control protocols
• MiNET, proprietary protocol from Mitel
• IAX, the Inter-Asterisk eXchange protocol used by the Asterisk open source PBX server and associated client software

Several different speech codecs can be used for stream audio compression. Commonly used codecs for VoIP traffic include G.711 and G.729, both ITU-T-specified codecs.