Archive for the ‘voip’ tag
ONT Notes – AutoQoS
- AutoQoS benefits
- Automates QoS for most deployments
- Protects business-critical apps to maximize availability
- Simplifies QoS deployments
- Reduces configuration errors
- Cheaper, faster, and simpler deployments
- Follows DiffServ
- Allows complete control over QoS configs
- Allows modification of auto-generated configs
- AutoQoS phases of evolution
- AutoQoS VOIP – Early version that configures the basics without discovery
- AutoQoS for Enterprise – Second version that only runs on routers and uses two-step process
- Autodiscovery using NBAR
- Generation of class maps
- AutoQoS key elements
- Application classification
- Policy generation
- Configuration
- Monitoring and reporting
- Consistency
- Interfaces that you can configure AutoQoS on
- Serial ifs with PPP and HDLC
- FR point-to-point subifs (NOT multipoint)
- ATM point-to-point subifs
- FR-to-ATM links
- Prerequsites
- No Qos policy already configured on if
- CEF enabled on if
- Correct bandwidth configured on if
- IP address on low-speed if
- Configuring AutoQoS Enterprise on a router (NOT a switch)
- auto qos discovery – begins discovery process
- auto qos – generates and applies MQC-based policies
- Configuring AutoQoS VOIP
- auto qos voip [ trust | cisco-phone ]
- Verifying AutoQoS on router
- show auto discovery qos – get autodiscovery results
- show auto qos – examine configuration generated
- Number of classes
- Classification options
- Marking options
- Queuing mechanisms
- Other QoS mechanisms
- If, subif, PVC where policy is applied
- show policy-map interface – look at if stats
- Verify AutoQoS VOIP
- show auto qos
- show policy-map interface
- show mls qos maps – shows CoS to DSCP mappings
- Possible issues with AutoQoS
- Too many traffic classes – manually consolidate some
- Configuration doesn’t change – rerun AutoQoS
- Configuration may not fit your situation – fine-tune it by hand
- Fine-tuning AutoQoS
- Use QPM
- CLI
- copy policy into editor, change, reapply
- AutoQoS can match on characteristics besides ACLs and NBAR
- match input interface
- match cos
- match ip precedence
- match ip dscp
- match ip rtp
ONT Notes – Pre-classify and End-to-end QoS
- VPNs (Didn’t ISCW cover this?)
- Provide
- Confidentiality
- Integrity
- Authentication
- Types
- Remote-access
- Client-initiated
- NAS-initiated
- Site-to-site
- LAN-to-LAN
- Extranet
- Remote-access
- Provide
- L3 Tunneling protocols
- GRE
- IPSec
- Pre-classify allows traffic to be classified before being sent across a tunnel or crypto-ed.
- qos pre-classify
- Provides a view into the original IP headers
- To classify on pre-tunnel header, apply the policy to the tunnel interface WITHOUT pre-classify.
- To classify on post-tunnel header, apply the policy to the physical interface WITHOUT pre-classify.
- To classify on pre-tunnel header, apply the policy to the physical interface WITH pre-classify.
- SLA – agreement with provider to guarantee QoS mechanisms across their network based on your markings.
- Assures availability, loss, throughput, delay, and jitter.
- End-to-end QoS
- To be effective, each hop in the path must have QoS configured similarly.
- Necessary in three locations
- Campus – within the customer network
- The edges – customer facing the provider, provider facing customer
- On the provider network
- QoS tasks
- Campus access switches
- Speed/duplex settings
- Classification
- Trust
- Phone/access switch configs
- Multiple queues on switch ports, including priority for VOIP
- Campus distribution
- L3 policing and marking
- Multiple queues on switch ports, including priority for VOIP
- WRED
- WAN edge
- SLA definitions
- LLQ
- LFI
- WRED
- Shaping
- Provider cloud
- Capacity planning
- PHB
- LLQ
- WRED
- Campus access switches
- Enterprise campus QoS implementation
- Implement multiple queues to avoid congestion
- Assign VOIP and video to highest priority queue
- Esablish trust boundaries
- Use policing to rate-limit excess traffic
- Use hardware QoS when possible
- Control Plane Policing (CoPP)
- Applies QoS policy to traffic destined for the router
- Routing protocols
- Management protocols
- Can be used to avoid DOS attacks
- Applied to control-plane in global config
- Applies QoS policy to traffic destined for the router
ONT Notes – Congestion Avoidance, Policing, Shaping, and Link Efficiency
- Tail drop drawbacks
- TCP synchronization – Dropping TCP packets from different flows can cause them all to window down and back up again at the same time in cycles.
- TCP starvation – Non-TCP or aggressive flows can starve everyone else out when TCP throttles back.
- No differentiated drop – Tail drop doesn’t care who you are, so you get dropped if the queue is full.
- RED – Random Early Detection
- Avoids tail drop by randomly dropping packets from the queue before it gets full
- Only dropped TCP flows slow down instead of everyone who has sent a packet since the queue filled
- Queues are smaller.
- Link utilization is more efficient
- Configured with
- Minimum threshold – start dropping when the queue is this size
- Maximum threshold – if the queue is this big, start tail dropping
- Mark probability denominator (MPD) – 1/MPD is the ratio of packets to drop when between the thresholds
- WRED – Weighted RED
- Based on IP precedence or DSCP values
- Less-important packets are dropped more aggressively than important packets
- Applied to an interface, VC or a class within a policy map
- CBWRED – Class based WRED
- Configured with CBWFQ
- Policing
- Limits subrate bandwidth (give you 100kbps on a T1)
- Limits traffic of certain applications
- Any traffic that exceeds police is dropped or re-classified; it’s a hard limit
- Inbound or outbound
- Shaping
- Sets a limit but buffers any in excess
- Requires memory to store the buffer
- Buffers = delay and/or jitter
- Outbound only
- Can respond to network signals like BECNs and FECNs
- Token and bucket
- The queue is a bucket; if a byte of data needs to be sent, it needs a token.
- If there are enough tokens, the traffic is considered conforming.
- If there aren’t enough tokens, the traffic is considered exceeding, which triggers the drop (policing), re-classify (policing), or buffer (shaping).
- Frame relay traffic shaping (FRTS)
- Only controls frame relay traffic
- Applied on subif or DLCI
- Support fragmentation and interleaving
- Reacts to FECNs and BECNs
- Compression
- Removed redundancy and patterns in data
- Less data = less latency
- Hardware compression or hardware-assisted compression does not involve the main CPU
- Software compression does
- Payload compression
- Header compression
- Link fragmentation and interleaving
- Small data might be waiting for larger data pieces to finish sending
- Chunks data into smaller fragments so they don’t have to wait
- Interleaving shuffles flows in the Tx queue
ONT Notes – Queuing
Here are some more notes from my studies. Of course, no one cares about them but me, but it’s my blog. I’m sure someone will find it useful. Please help to correct dumbass mistakes.
- Congestion
- Speed mismatch – traffic leaves a lower-bandwidth interface than the one it came in on
- Aggregation problem – lots of links with one egress of equal bandwidth
- Confluence problem – a bunch of traffic needs to egress out of the same interface
- Queuing
- Transmit queue (TxQ) – hardware queue; there’s only one you can’t touch
- Software queue – where packets wait to be sent; there are many queue-types that you modified to police traffic
- FIFO
- If I beat you to the router, I leave the router first.
- Possible long delays, jitter, and starvation
- Priority queuing (PQ)
- Four queues
- High-priority
- Medium-priority
- Normal-priority
- Low-priority
- Scheduler starts from high and work to low
- When the high queue is empty, it processes a packet from medium, then starts all over
- Can you say starvation?
- Four queues
- Round robin queuing (RR)
- One packet from this queue, one from the next, etc., then start over again
- Custom queuing (CQ)
- Weighted round robin
- Queues are given weights (bandwidth guarantees)
- Weighted Fair Queuing (WFQ)
- Default queuing on slow links ( < E1 )
- Divides traffic into flows
- Equal bandwidth is given to each flow
- Provides faster scheduling to low-volume flows
- Provides more bandwidth to higher-priority flows
- Flows identified by a hash
- Source IP
- Destination IP
- Protocol number
- ToS
- Source port
- Destination port
- Each unique has is a new flow
- No way to allocate bandwidth among the flows
- By default, up to 256 queues are made, but that is changeable to a power of 2 between 16 and 4096
- If the max number of flows is reached, queues are reused for other flows
- If a queue is full, a packet may be dropped.
- WFQ early dropping drops packets when the queue reaches the congestive discard threshold (CDT)
- Advantages
- Simple configuration
- No starvation
- Guarantee processing of all flows
- Drops packets from big-hitter flows
- Faster service no low-hitters (interactive) flows
- Standard on (nearly) all IOS devices
- Disadvantages
- Classification and scheduling are not configurable
- Only on slow links
- No guarantee of bandwidth or delay
- Class-based Weighted Fair Queuing (CBWFQ)
- User-defined queues for flexibility
- Configured with class-maps via MQC
- Weights are calculated based on values give in class-map
- Bandwidth – guarantee this much bandwidth
- Bandwidth percent – give me this much of the available bandwidth
- Bandwidth remaining percent
- Advantages
- User-defined traffic classes
- Each queue gets its own bandwidth
- Scalability
- Disadvantages
- No delay guarantee (not good for real-time application like voice)
- Configuring
class-map TESTCM1 match access-group 100 ! class-map TESTCM2 match access-group 200 ! policy-map TESTPM class TESTCM1 bandwidth 64 class TESTCM2 bandwidth 128
- Low-latency Queuing
- Includes strict priority queue for delay-sensitive data
- Strict priority queue is policed to avoid starvation of other queues
- Configured the same way as normal CBWFQ, but with the priority keyword
- This configuration makes TESTCM2 a priority queue
class-map TESTCM1 match access-group 100 ! class-map TESTCM2 match access-group 200 ! policy-map TESTPM class TESTCM1 bandwidth 64 class TESTCM2 priority bandwidth 128
ONT Notes – Classification, Marking, and NBAR
Here’s another set of notes from my ONT studies. I’m sure someone will find it useful. Please help to correct dumbass mistakes.
- Classification is done with traffic desriptors
- Ingress interface
- CoS value on ISL or 802.1P frames
- Source/destination IP address
- IP Precedence or DSCP value
- MPLS EXP
- Application type
- Layer 3 QoS
- Type of Service (ToS) is 8-bit field.
- First 3 bits of ToS are the IP precedence.
- First 6 bits of ToS are the DSCP value.
- Last 2 bits of ToS are explicit congestion notification (ECN).
- Layer 2 QoS
- Ethernet
- Class of Service (CoS)
- On 802.1P frame
- 3-bit priority (PRI) field
- 000 – Routine – Best-effort
- 001 – Priority – Medium priority
- 010 – Immediate – High priority
- 011 – Flash – Call signaling
- 100 – Flash-Override – Video conferencing
- 101 – Critical – Voice bearer
- 110 – Internet – Reserved
- 111 – Network – Reserved
- Frame Relay
- 1-bit discard eligible (DE) field
- ATM
- 1-bit cell loss priority (CLP) field
- MPLS (layer 2 1/2)
- 3-bit experimental (EXP) field
- By default, the 3 most significant ToS bits (IP Precedence bits) are copied to EXP
- Ethernet
- Per-hop Behavior (PHB)
- “an externally observable fowarding behavior of a network node toward a group of IP packets that have the same DSCP value”
- In other words, treat packets with the same DSCP value in the same manner – scheduling, queuing, policing, etc.
- Behavior aggregate (BA) is a group of packets with the same DSCP value
- DSCP
- DSCP is chopped up into 4 PHBs
- Class selector PHB – (000) old IP precedence compatibility
- Default PHB – (000) best effort
- Assured forwarding (AF) PHB – (001, 010, 011, 100) guarantee bandwidth
- Provides 4 queues for 4 classes of traffic (AF1-4)
- Also specifies drop preference (ex., AF41, A13) where second number is preference (higher is more probable to be dropped)
- Each queue must have (W)RED to avoid drops
- No queue is any better than the other
- Backward compatible with IP precedence
-
- Expedited forwarding (EF) PHB – (101) low delay
- Minimum delay
- Bandwidth guarantee
- Policing
- Expedited forwarding (EF) PHB – (101) low delay
- DSCP is chopped up into 4 PHBs
- Trust boundaries
- Establish DSCP values as close to the source as possible
- On the device (IP phone), access switch, or distribution switch
- The core should never assign DSCP values
- Only trust DSCP values from devices you trust
- Examine and rewrite values from untrust sources
- Establish DSCP values as close to the source as possible
- Network-based Application Recognition (NBAR)
- Protocol discovery – discovers what protocols you’re running on your network
- Traffic statistics collection – keeps tracks of stats on each protocol
- Traffic classification – NBAR protocols can be used in class-maps to define traffic to be services
- Packet description language models (PDLMs) – table of what protocols NBAR recognizes
- Limitations
- Doesn’t work on EtherChannel interfaces
- Only handles 24 URLs, hosts, or MIME types
- Only analyzes first 400 bytes of the packets
- Requires CEF
- Doesn’t work on HTTPS, multicasts, or fragments
- Ignored traffic destined for the router itself
- NBAR commands
- Router(config)# ip nbar pdlm pdlm-name : Update the PDLM table
- Router(config)# ip nbar port-map protocol-name [tcp|udp] port-number : Adds an entry to the PDLM table
- Router# show ip nbar port-map protocol-name : Shows what’s in the PDLM table
- Router# show ip nbar protocol-discovery : Shows what’s been discovered
- Router(config-cmap)# match protocol name : a class-map match for an NBAR-discovered protocol
- Special protocol matching
- Can match beyond the port number with deep packet inspection
- Matches HTTP hostname, URL, or MIME type
- Matches fast-track P2P
- Matches RTP content
ONT Notes – Intro to QoS
I’ll try to keep it a little shorter this time.
Major issues for converged enterprise networks
- Available bandwidth: competition among applications
- Fixes
- Increase bandwidth: More power!
- Properly queue based on classification and marking: QoS
- Compress: cRTP, TCP header compression, etc.
- Fixes
- Delay: Lead time to get a packet to the destination
- Types of delay
- Processing delay: routing, switch delay
- Queuing delay: how long a frame stays in an output queue
- Serialization delay: how long to put the frame on the wire
- Propagation delay: the time to cross the physical medium
- Types of delay
- Jitter (delay variation): Variation is the delay
- Different delays mean different arrival times
- De-jitter buffers save up packets to reduce jitter (like the old CD writers)
- Fixes
- More bandwidth
- Prioritize sensitive data and forward first
- Remark (reclassify) packets based on sensitivity
- Enable L2 payload compression: make sure compression delay isn’t worse than the jitter
- Use header compression
- Packet loss: Packets are lost in the network somewhere
- Fixes
- More bandwidth
- Increase buffers space: more room for the queue on the interface
- Provide guaranteed bandwidth: Queuing and QoS
- Congestion avoidance
- Random Early Detection (RED) and weighted RED (WRED) drop packets before the queue is full
- Selective dropping is better than FIFO or LIFO dropping
- Fixes
QoS History
- Priority queuing: gives certain data the right-of-way for transmission
- Weighted Fair Queuing (WFQ): prevents small packets from waiting too long for big packets
- RTP priority queuing: Gives voice packets the right-of-way
- CAC: Makes sure we don’t fill up the queue or pipe with voice traffic
Implementing QoS
- Step 1: Identify traffic types and requirements
- Network audit
- Business audit
- Define bandwidth requirements for each class found
- Step 2: Classify the traffic
- Common classes
- VOIP
- Mission-critical
- Signal traffic: for VOIP
- Transactional application: SAP, ERP
- Best-effort: Everything else
- Scavenger: Crap you don’t care about like P2P and your boss’s email
- Common classes
- Step 3: Define policies for each class
- Tasks for each class
- Set max bandwidth
- Set min bandwidth
- Assign relative priorities
- Apply congestion avoidance, congestion management, etc.
- Tasks for each class
QoS Models
- Best-effort: no QoS
- Scalable
- Easy
- No service guarantee: doesn’t care what you’re trying to do
- No service differentiation: all traffic is equal
- Integrated Service (IntServ)
- Hard-QoS
- Uses RSVP to guarantee bandwidth through the entire path
- Requires
- Admission control
- Classification
- Polices the traffic (ceiling)
- Queuing
- Scheduling
- Advantages
- End-to-end resource admission control
- Per-request policy admission control
- Signaling of dynamic ports
- Disadvantages
- Continuous signaling
- Not scalable
- Differentiated Services (DiffServ)
- Soft-QoS
- Configured on each hop
- Traffic is classified
- Enforces different treatment on different classes
- Defined based on business requirements
- Benefits
- Scalable
- Supports lots of service levels
- Drawbacks
- No absolute guarantee of service
- Complex configuration throughout network
QoS Implementation Methods
- CLI
- Old school
- Not used any more
- Modules QoS CLI (MQC)
- Step 1: class-map
- Step 2: policy-map
- Step 3: service-policy
- AutoQoS
- Automatically generates classes and policies based on traffic it sees
- Super-simple
- Requires CEF, NBAR, and correct bandwidth statements
- SDM QoS Wizard
- Next, next, next
- Can be used to implement, monitor, or troubleshoot QoS
ONT Notes – VOIP Networks
Here are some of the notes I’ve been taking while reading over the ONT book. I hope it benefits somebody. Feel free to correct any stupid mistakes as a paraphrase to avoid a lawsuit.
There’s way too much info here. I’ll refine the process a little better for the next topics.
Benefits of Packet Telephony Networks
- More efficient use of bandwidth and equipment – Packet telephony networks don’t dedicate channels or a static bandwidth to a call; it’s just another network application.
- Consolidate network expense – The common infrastructure (IP-based networks) keeps you from having to support another distinct network for voice like in traditional PBX implementations.
- Improved employee productivity – The phone can be used for more than just phone calls by utilizing the XML interface to run applications or provide content from the network.
- Access to new communications devices – IP phones can communicate with computers, network gear, PDAs, etc., and not just the PBX.
Packet Telephony Components
- Phones – These include analog phone, digital phones, IP phones, softphones, etc.
- Gateways – These devices connect the different devices that cannot access the IP network. For example, making a 911 call from your IP phone requires a gateway that switches and converts your VOIP conversation to the PSTN.
- Gatekeepers – These are devices that handle call routing (resolving an IP to an extension/phone number) and call admission control (CAC, grants permission to make the call).
- Multipoint control units (MCUs) – These are conference bridges that connect a bunch of streams together and present it to all participants. Some can do video as well.
- Call agents – These are devices used in a centralized model that handle the call routing, address translation, call setup, call maintenance, and call termination.
- Application and database servers – These provide required and optional services to the packet telephony network and include TFTP servers for configuration and OS download and XML servers for application use.
- Digital signal processors (DSPs) – These guys converts signals from one form to another. They convert analog to digital signals, digital to packetized data in the form of a codec, from codec to codec, etc.
Analog Interfaces
- Foreign Exchange Office (FXO) – These are interfaces that expect to connect to a CO or equivalent. You connect these to your wall jack to get access to the PSTN.
- Foreign Exchange Station (FXS) – You connect your analog devices (phones, modems, faxes, etc.) to these guys to get dial tone.
- Ear and Mouth (E&M) – These are the old-school way to connect PBXes together.
Digital Interfaces
- Basic Rate ISDN (BRI) – These give you 2 64kbps channels (bearer channels) to run voice over. It also includes a 16kbps D (delta) channel with 48kbps of framing overhead to give you 192kbps.
- T1 (North America) – This is a channelized T1 or a Primary Rate ISDN (PRI).
- Common Channel Signaling (CCS) – The D channel is dedicated to signaling, giving you 23 64kbps channels.
- Channel Associated Signaling (CAS) – There is no D channel, but every bearer channel dedicates a few data bits for its own signaling.
-
- E1 (North America) – This is a channelized E1 or a Primary Rate ISDN (PRI).
- Common Channel Signaling (CCS) – The D channel is dedicated to signaling, giving you 30 64kbps channels.
- Channel Associated Signaling (CAS) – There is still a dedicated D channel, so you still have 30 64kbps channels to use.
- E1 (North America) – This is a channelized E1 or a Primary Rate ISDN (PRI).
VOIP Signaling
- H323. – ITU Standard that uses a whole mess of RFCs; distributed model
- Media Gateway Control Protocol (MGCP) – IETF RFC 3435; centralized model
- Session Initiation Protocol (SIP) – IETF standard; distributed model
Phone Call Stages
- Call setup – connects the call between the endpoints
- Call routing – figures out where the call is going
- CAC (optional) – Do you have enough resources (i.e., an available channel or bandwidth) to make the call?
- Call negotiation – negotiates the source and destination IPs, source and destination UDP ports, and codec.
- Call maintenance – collects call statistics for on-demand or historical use
- Call teardown – hanging up and terminating the connection
Digitizing Analog Signals
- Sampling – Periodic capturing and recording of voice resulting in a pulse amplitude modulation (PAM) signal
- Quantization – Assigning numerical values to the PAM signal
- Encoding – Converting the quantization to binary
- Compression (optional) – compressing the binary stream
- Pulse code modulation (PCM) converts analog to digital, but it doesn’t use compression. It takes 8000 samples per second and converts each sample to an 8-bit number, giving 64kbps of capacity.
Digital to Analog
- Decompression (optional)
- Decoding and filtering – binary is converted back to a PAM signal; filtering removes any noise from the conversion
- Reconstructing the analog signal
The Nyquist Theorem
- The number of samples required to accurately encode (and decode) a signal is twice the highest frequency of the signal.
- Since telephone lines can only transmit up to 3400 Hz (4000 Hz for simplicity), the sample rate should be 8000 samples/second.
Measuring Compression Qualities
- Mean opinion score (MOS) – ITU standard technique for measuring quality of codec; subjective score from 1 to 5
- Perceptual speech quality measurement (PSQM) – Another ITU standard technique for measuring quality of codec; test equipment score from 0. to 6.5
- Perceptual analysis measurement system (PAMS) – Developed by BT; predictive system
- Perceptual evaluation of speech quality (PESQ) – Another ITU standard; combines PSQM and PAMS; objective measurement of factors including subjective values
Digital Signal Processors (DSPs)
- Provide 3 major services – voice termination, transcoding, conferencing
- Also performs compression (codec), echo cancellation, voice activity detection (VAD), comfort noise generation (CNG), and jitter handling
- Conferencing among participants with the same codec is called a single-mode conference.
- Conferencing among participants with different codecs is called a mixed-mode conference.
Protocols
- VOIP calls run over Real Time Protocol (RTP).
- RTP provides sequence reordering, time-stamping, and multiplexing
- Rides on UDP ports 16384-32767
- Voice does not need the reliability (retransmission) of TCP since retransmitted data is no longer useful (I already said that).
- VOIP packets headers:
- IP – 20 bytes
- UDP – 8 bytes
- RTP – 12 bytes
- L2 headers vary depending on technology (Ethernet = 12 bytes, MPLS, etc.)
- 2 10-ms packages are usually in one packet (20ms of voice)
- G.711 (64kbps) produces 160 bytes from 20 ms of voice.
- G.729 (8kbps) produces 20 bytes from 20 ms of voice.
cRTP
- Compressed RTP (cRTP) reduces the headers
- After the first packet lands, the IP, UDP, and RTP headers won’t change, so why send them again?
- The headers are reduced to a hash.
- cRTP reduces the headers to 4 bytes with a UDP checksum and 2 bytes without a UDP checksum.
- Slow links only
- Processing overhead
- Finite delay in packetization
Packet Size Effect on Bandwidth
- The size of a voice frame depends on:
- Packet rate and packetization size – rate is inversely proporational to size
- IP overhead – RTP, UDP, IP, cRTP overhead
- L2 overhead -
- Tunneling overhead – IPSec, GRP, MPLS, etc.
- Codecs have different bandwidth
- G.711 (PCM) – 8000 samples per second @ 8 bits per sample = 64 kbps
- G.726 (Adaptive Differencial PCM – ADPCM) – Variable bit rate of 32 kbps, 24 kbps, or 16 kbps
- G.722 (Wideband Speech Encoding) – 2 subbands using modified ADPCM of 64 kpbs, 56kbps, or 48 kbps
- G.728
- G.729 – 10 samples per 10-bit code = 8 kbps
Calculating Total Bandwidth
- Step 1 – Determine codec and packetization period: What does the codec require in bandwidth? How many samples per packet (usually 2)?
- Step 2 – Determine link-specific overhead: Encapsulation? cRTP?
- Step 3 – Calculate packetization size: Size of voice payload; codec bandwidth * packetization period / 8 = voice payload in bytes
- Step 4 – Calculate total frame size: IP + UDP + RTP + Tunneling + data link + packetization size
- Step 5 – Calculate packet rate: 1 / packetization period (ex., 20ms packetization period is 1/0.020 = 50 packets per second)
- Step 6 – Calculate total bandwidth: Total frame size * packet rate
VAD and Bandwidth
- Common for 1/3 of conversation to be silence
- VAD bandwidth savings depends on:
- Type of audio: regular phone call (two-way), conf call (one-way), music on hold (MOH)
- Background noise: noise may be detected as voice
- Other factors: language, culture may influence amount of silence
Enterprise VOIP Implementations
- Consists of gateways, gatekeepers, Cisco Unified CallManagers (CCM), Cisco IP Phones
- Routers can provide the voice gateway function by connecting the IP network to the WAN (and other gateways), PSTN, PBXes, etc.
- Survivable Remote Site Telephony (SRST) allows local calling and use of PSTN while services are down
Functions of CCM
- Call processing – routing, signaling, accounting
- Dial plan administration - call routing
- Signaling and device control – configuration and instruction in case of events
- Phone feature administration – button programming, profiles, etc.
- Directory and XML
- API for interface – allows custom programming for IP phones
Enterprise Deployment Models
- Single-site: You have one site, and everything is there.
- Multisite with centralized call processing: You have multiple sites, but the main site has the CCM cluster.
- Multisite with distributed call processing: You have multiple sites, and each site has its own CCM cluster.
- Clustering over WAN: You have multiple sites, and each site has a part of one big CCM cluster.
IOS Voice Commands
----- R1 ----- ! FXS on 1/1/2 Dial-peer voice 1 POTS destination-pattern 120 port 1/1/2 ! Extension 230 is on R2 Dial-peer voice 2 R2 destination-pattern 230 session target ipv4:10.1.1.2 ----- R2 ----- ! FXS on 2/2/1 Dial-peer voice 1 POTS destination-pattern 230 port 2/2/1 ! Extension Dial-peer voice 2 R2 destination-pattern 120 session target ipv4:10.1.1.1
Call Admission Control (CAC)
- QoS can guarantee bandwidth but can only reserve so much (say, for 2 simultaneous calls).
- CAC make sure that resources are available (denies a new call if 2 calls are already placed).
- Dropped packets affect every call – not just the new ones
—–
Additional Reading
- H.323 Sources on Wikipedia
- MGCP – RFC 3435
- SIP – RFC 3261
- Nyquist Theorem on Wikipedia
- MPLS on Wikipedia
Cisco IP Phone Videos at Blindhog.net
Josh over at Blindhog.net has an article linking to a bunch of Cisco IP Phone videos — from the 7906 to the 7975. These are Cisco videos and a good place to start if you don’t know anything about their IP phones.