VoIP

Technical Components of a Call

Call signalling, media protocol, and codec.

• Call signaling/control
  • SIP or H.323
  • SIP media identification and negotiation used Session Description Protocol (SDP)
  • SIP can be encrypted with TLS SIP is primarily used in setting up and tearing down voice or video calls. It also allows modification of existing calls. The modification can involve changing addresses or ports, inviting more participants, and adding or deleting media streams.
• Media protocol (real-time protocol)
  • RTP (IETF RFC 3550) or RTCP
  • Typically on top of UDP
  • Secure RTP provides encryption (RFC 3711)
• Codec
  • Many (IETF RFC 3551). G.711u, G.726, etc.

Gateways and device control protocols

Additionally, H.323 and SIP have complementary device control protocols: H.248 and MGCP. These involve gateways.

A gateway has an IP interface on one side, and a legacy phone interface on the other. This is called a media gateway (MG). Originally, such a gateway handled both the IP to PSTN interface and call control, but the call control functions have been split off to a media gateway controller (MGC or Call Agent).

Since the MC and MGC functions are typically split these days, H.248 or MGCP acts as the protocol between the MG and the MGC.

     MG = IP to PSTN gateway
              |
         H.248 or MGCP
              |
      MGC/CA doing call control

A call

1. ADC converts analog voice to digital bits (in VoIP phone)
2. Bit are compressed, e.g. G.711, G.722, or GSM (in VoIP phone)
3. Insert packets of voice bit into a real-time protocol
4. Call signaling
5. VoIP gateway
   • Either IP or PSTN 1.
6. Convert bit back to analog voice

Links

https://www.packetizer.com/ipmc/papers/understanding_voip/voip_protocols.html https://en.wikipedia.org/wiki/Session_Initiation_Protocol