VOP

Voice over Packet
Basic Understanding
Nayan Gaywala 1
Email:nayan@pacbell.net
Overview
• VOIP Scenarios • Fax Relay
• Voice compression
• Echo Canceller
• DTMF Relay
• Jitter Buffer
• Voice Quality
• Session Layer
• Caller Id
Nayan Gaywala 2
Cheap Long Distance
Dial a local service provider # followed by

a destination country code & phone #
Local call Local call
USA India
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Home User
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Office Environment
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DLC
• Dial tone comes from the Switch

• Hook, Ring on/off sent as RTP NTE
• DTMF detection on the Switch
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Encapsulation
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Voice Compression at its core
• H(z) filter parameters, pitch period, gain and

voiced/unvoiced switch constitutes compression.
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Why Echo a problem in VOIP
• Far-end hybrid echo is a problem. In VOP, delay in the call

path is not distance related. It is this delay which makes
echo a particular problem in packet telephony.
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Echo
• Local hybrid(near-end) echo appears as side-tone which is
desirable.
• Annoyance level increases with both strength and delay.
• Your echo canceller helps improve the quality at the
remote end.
• G.168 specifies tests to characterize the EC
implementation.
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Echo
ACANC Send path ANLP
Network Sin Non-linear Sout

Elements Digital Subtractor
processor
LRET
Near-end LRES
AECHO
Hybrid Far-end
Echo estimator and
other control circuitry LRin
Network
Elements Rout Rin
Receive path
T1524860-96
• NLP suppresses the residual echo and inserts comfort noise

• Double-Talk detection is needed to freeze the adaptation
when both ends are talking.
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Why need DTMF Relay
• Parametric vocoders like G.729 & G.723 exploit the
features of speech to accomplish compression. Non-speech
signal like DTMF when they pass through the
coding/decoding cycle of a parametric vocoder are not
reproduced faithfully.
• The idea is to detect local tones, relay tone info bypassing
the vocoder and re-generate them at the remote end.
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DTMF Relay
• In-band handled just like voice i.e. no relay
• Out-of-band i.e. using relay.
– Two special RTP packet types as defined in RFC2833.
• Events
• Tone
– Using session level signaling. For example:
• In H.323 carried by H.245 User Indication message
• In H.248 carried by Notify message
• Q.24 specifies the requirements for the DTMF detector.
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Jitter Buffer
• Smoothen out jitter introduced by packet network.
• Trade off : Delay vs. acceptable under-run/overrun.
• Fills to the nominal jitter level before the playout begins.
• Packets dropped after the max jitter level.
• Fixed vs. Adaptive implementation
– FJB: headstart is constant at nominal jitter level
– AJB: headstart varies between nominal jitter and max jitter setting,
but for FJB doesn’t change
– AJB: headstart adaptation generates distortion
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Jitter Buffer
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Jitter Calculation
• First the difference D(i,j)=(Rj-Ri)-(Sj-Si)=(Rj-Sj)-(Ri-Si)
in transit time between two packets is calculated.
• This difference is smoothed out to estimate the Inter-
arrival jitter J(i)=|D(i-1,i)|/16+15/16*J(i-1). Smoothening
1
filter is a single pole low-pass filter H ( z ) 16
15 1
1 z
16
• Jitter estimate is used to adjust the adaptive jitter buffer

and send RTCP reports.
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Factors affecting Overall Quality
• End-to-end Delay
– Algorithmic delay. Vocoder encodes frame length worth of
samples. 1 sample for G.711 while 30ms for G.723.
– Packetization delay. Sometimes RTP Packetization period multiple
of the vocoder frame length. Typical periods are 10, 20 & 30ms.
– Processing delay. Depends on how fast we encode/decode the
frame. Has to be less than the frame length for real-time
implementation.
– Propagation delay. This is the absolute network propagation delay.
– Jitter buffer length.
– Buffering delay. Depends on implementation.
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• Compression Loss
– Loss of voice quality through a compression-decompression cycle.
• Echo
– talker’s own voice reflected back over the network.
• Silence Suppression – VAD/CNG
– Clipping of voiced parts of the speech. Effects of artificial comfort
noise.
• Loudness level
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• Network Impairments
– Jitter - Inter arrival time variation
– Packet loss
– Absolute delay – Network propagation delay
– Reorder
– Duplicate
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Typical H.248 call
Off hook
Exchange codec, RTP

port frame length info
Disconnect the call
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Typical SIP call
Send collected digits to server
along with SDP info(codec, RTP
port frame length)
Answer side ringing
Answer side SDP info
Disconnect the call
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Caller Id
• In VoIP we are concerned with CID generation.
• Data sent using FSK (Bell 202) at physical layer
• Two flavors of CID
– On-hook (Type 1). Plays CID between the first and second ring
– Off-hook (Type 2). Plays during the call
• Two formats
– Single Data Message Format (SDMF)
– Multiple Data Message Format (MDMF)
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Type 1 Caller Id
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Type 2 Caller Id
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Various Answer tones
Symbol Spec Purpose
ANS/ Fax CED Pure 2100Hz Used by voiceband modems like

V.22bis or Called Fax M/C to
disable G.164 Echo Suppressor
/ANS 2100Hz with phase Used by voiceband modems like
reversals V.32bis to disable G.165/G.168
Network Echo Canceller
/ANSam 2100Hz with phase Used by V.34 to disable EC and
reversals and distinguish between V.34 and
amplitude modulated V.32 modem
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VOP

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VOP

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Voice over Packet

Dial a local service provider # followed by

Local call Local call

• Dial tone comes from the Switch

• H(z) filter parameters, pitch period, gain and

• Far-end hybrid echo is a problem. In VOP, delay in the call

Network Sin Non-linear Sout

• NLP suppresses the residual echo and inserts comfort noise

• Jitter estimate is used to adjust the adaptive jitter buffer

Exchange codec, RTP

Disconnect the call

Answer side SDP info

Disconnect the call

ANS/ Fax CED Pure 2100Hz Used by voiceband modems like

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