Section: Advanced Addressing and Routing Solutions

Bandwidth provisioning for an IP telephony call consists of the voice stream traffic and the call control traffic.
These elements are unique to an IP telephony call.
The network infrastructure should be examined to see if the required bandwidth exists to support the voice and call-control applications. The sum of the bandwidth necessary for each major application, including voice, video, and data, should not exceed 75% of the total available bandwidth for each link. Voice traffic can be characterized as:
Smooth
Benign
Drop sensitive
Delay sensitive
Voice packets are typically around 60 to 120 bytes in size. For good voice quality, packet loss should be less than 1 percent and delay should be no more than 150 ms.
The IP telephony voice call-control procedures also generate traffic. The call control procedures are in the areas of call setup, maintenance, redirect, and tear down. There are special protocols such as H.323 and Media Gateway Control Protocol (MGCP) that handle these procedures.
Voice applications are delay-sensitive. Speech is sampled by voice processors referred to as a codec (coder/ decoder). Then the digitized voice-sample outputs of the codecs are sent into the network towards the receiver at regular intervals in real-time transport protocol (RTP) packets. If these packets containing the voice samples are delayed for any reason behind other data traffic, the quality of the voice conversation suffers.
The transportation of these voice applications in RTP packets through the IP network handled by H.323 protocols and devices is referred to as Voice over IP (or VoIP for short).
The following are other network and design considerations besides bandwidth relating to IP telephony infrastructure support:
Determine if the cabling plant can support the IP telephony equipment.
Determine if the switch hardware can supply power to attached IP telephony equipment or if additional hardware is required.
Ensure that infrastructure supports priority end-to-end VLANs and QoS networking.
Internet Group Management Protocol (IGMP) is used for managing the membership of IP multicast groups and is not an element unique to an IP telephony call.
Routing protocol packets (RIP, OSPF, and EIGRP) are used by routers to share routing information, and are not elements unique to an IP telephony call.
The speed of the segment to the telephone is important to VoIP, but that is not an element unique to an IP telephony call.
Objective:
Layer 2 Technologies
Sub-Objective:
Configure and verify VLANs
References:
Cisco > Support > Technology Support > Voice > Telephony Signaling