Centralized Call Processing (VoIP Bandwidth and CAC)

Inter-site IP phone calls in the centralized call processing model are routed over the wide area network (WAN). To preserve valuable WAN bandwidth, most organizations compress phone calls with the G.729 or iLBC audio codecs. G.729 compresses audio calls to 8kbps + 16kbps (IP/UDP/RTP encaptulstion) for a total of 24kbps per call. The iLBC audio codec uses 15.2kbps + 16kpbs = 31.2kbps for the same phone call. Although iLBC uses almost twice the codec bandwidth of G.729, iLBC has a higher MoS and a better degraded mean opinion score (DMoS) than G.729. Cisco Type B phones running on CUCM 6.0 or later, support the iLBC and G.722 high-fidelity audio codecs (7970, 79x1, 79x2, 79x5, 7906). The G.722 codec is used for phone call within each site, but a compressed audio codec is normally used to route calls between sites. The 7970, 79x1, and 7906 phones did not ship with the high-fidelity handsets required to hear the 16khz frequency response range supported with the G.722 audio codec. G.722 consumes 80kbps of bandwidth per call (64kbps audio codec + 16kbps overhead). The G.722 audio codec compresses high quality audio to the same 64kbps bandwidth as the G.711 audio codec. Cisco Unified Communications Manager (CUCM), Cisco Unified Communications Manager Express (CUCME) and VoIP dial peers in Cisco routers do not have any bandwidth restrictions by default. These systems could potentially route hundreds of phone call over a T-1 data WAN link, thereby degrading the quality of all the phone conversations. Call Admission Control (CAC) technology is used to artificially limit the number of phone calls between sites to ensure resources are not over utilized. The centralized call processing model uses the Call Manager locations configuration to logically bind Cisco phones to physical geographical locations. The locations configuration parameter is configured in the Call Manager administration System menu. G.729 calls are configured in increments of 24kbps and G.711 phone calls are configured in increments of 80kbps. The locations configuration should closely align to the priority queue (PQ) QoS configuration on the wide area network link. The PQ should take layer 2 encapsulation overhead into consideration, while the location configuration will not. The QoS SRND has tables with bandwidth values for both G.729 and G.711 with different layer 2 encapsulations. A link to the Enterprise QoS SRND is in the references section of this blog. Let’s examine the math behind one of the voice samples. If we examine a G.729 phone call with a sniffer, we can see that each voice sample is 20 bytes in size. Each Cisco Voice over IP phone and interface samples voice every 20 milliseconds by default. A millisecond is 1/1,000 of a second and 20/1,000 can be simplified as 1/50. A Cisco voice over IP phone interface generates 50 packets per second (pps). We will now figure out how many bits per second is used by G.729 by converting the packet into bits. 20 bytes * 8 bits / byte = 160 bits, and 160 bits * 50pps = 8000bps (8kbps). In the next blog, we will further investigate centralized call processing configuration options. REFERENCES: CUCM 7.x Solution Reference Network Design (SRND) Guide: http://www.cisco.com/en/US/netsol/ns818/networking_solutions_program_home.html www.cisco.com/go/srnd SRST 7.0 Administration Guide http://www.cisco.com/en/US/docs/voice_ip_comm/cusrst/admin/srst/configuration/guide/srstsa.html Wikipedia: Mean Opinion Score http://en.wikipedia.org/wiki/Mean_Opinion_Score Wikipedia: Internet Low Bandwidth Codec http://en.wikipedia.org/wiki/ILBC iLBC Degraded MOS (DMOS) Charts http://www.ietf.org/proceedings/02mar/slides/avt-6.pdf Cisco Press Voice Codec Comparisons http://www.ciscopress.com/articles/article.asp?p=606583&seqNum=4 QoS Solution Reference Network Design Guide http://www.cisco.com/en/US/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/QoS-SRND-Book.html Voice Over IP - Per Call Bandwidth Consumption http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml

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