This document discusses jitter, latency, and delay in network communications. It provides definitions and explanations of these terms: 1. Jitter is the variation in the delay of received packets caused by network congestion, queuing, or errors, rather than packets being transmitted at an even pace. This can cause gaps in audio if packets are missing. 2. Delay and latency refer to the time it takes a bit to be transmitted from source to destination. Jitter is a type of delay that varies over time. 3. Solutions to reduce jitter include increasing the receive jitter buffer size and delay, using larger RTP packets, and lowering audio quality.

1. Step Description Value (30 % HARQ)
1 eNBProcessingDelay(S1-U->Uu) 1 ms
2 Frame Alignment 1,022 ms
3 TTI for DL DATA PACKET 0,675 ms
4 HARQ Retransmission 0,3 * 5 ms
5 UE ProcessingDelay 1 ms
6 S1-U TransferDelayand aGW 7 ms (Note)
Total one way delay 12,2 ms
NOTE: The delaybudgetwasmeasuredinproductionenvironments,The S1-Udelayisbandwith
depended.
Speech pathdelay
Jitter is defined as a variation in the delay of received packets. The sending side
transmits packets in a continuous stream and spaces them evenly apart. Because of network
congestion, improper queuing, or configuration errors, the delay betweenpackets can vary
instead of remaining constant, as shown in the figure.
n the RRC you can increase jtter buffer size and delay. Try to use larger RTP package, test with
40ms. Lower audio quality = 0 can be a solution. That's about what you can do locally in the
RRCs
ncrease rx jitter buffer to min 12+
- increase rx jitter delay to min 10+
- audio packet size 20 or 40
Jitter is a variation in packet transit delay caused by queuing, contention and serialization
effects on the path through the network. In general, higher levels ofjitter are more likely to occur
on either slow or heavily congested links.

2. Delay and latency are similar terms that refer to the amount of time it takes a bit to be
transmitted from source to destination. Jitter is delay that varies over time. One way to
view latency is how long a system holds on to a packet. ... The speed of a system is affected by
congestion and delays.
Jitter in IP networks is the variation in the latency on a packet flow between two systems, when
some packets take longer to travel from one system to the other.Jitter results from network
congestion, timing drift and route changes
Handover interruption delay
HybridARQ enabledRate adaptationbasedonCQIfeedbackRLCAMmode Multiple bearersQoS
scheduler
Duringhandoverprocess,forsome period,userequipments cannotexchangeuserplane packetswith
any of the base stations.Thisperiodisknownashandoverinterruptiontime.Itincludesthe time
requiredtoexecute anyradioaccessnetworkprocedure,radioresourcecontrol signaling,orother
message exchanges betweenthe userequipmentandthe radioaccessnetwork.The impactof intraLTE-
Advancedhandoversoninterruptiontimeislessthanorequal to that providedbyhandoversinLTE.In
LTE-Advanced,sub-framesize,alsoknownasTransmissionTime Interval (TTI),of 1msmakesitcapable
of adaptingtofast changingradiolinkconditionsandallowsexploitationof multiuserdiversity[7].In
LTE-Advanced,processingdelaysindifferentnodesandRACHschedulingperiodare reducedin
comparisontoLTE. RACH cycle isdecreasedfrom5.0ms to1.0ms. Thistutorial outlinesthe procedures
involvedinhandoverprocessandanalyzesthe performance of handoverinterruptiontimeforbothFDD
and TDD modes.The paperis organizedasfollows:SectionIIexplainsthe minimumrequirementssetby
IMT-Advancedandassumptionsof analysis.SectionIII
presentsthe analysisof handoverinterruptiontimebeforethe conclusionsare drawninSectionIV.II.
REQUIREMENTS & ASSUMPTIONSA.MinimumRequirementsThe IMT-Advancedproposal shallbe able
to supporthandoverinterruptiontimesspecifiedinTable I[8].TABLE I IMT-A REQUIREMENTS Handover
Type InterruptionTime (ms) Intra-Frequency27.4Inter-FrequencyWithinaspectrumband40.0
Betweenspectrumbands60.0 B. AssumptionsHandoverInterruptiontime forintra-frequencyand
interfrequencyisthe same asitdoesnot dependonthe frequencyof the targetcell aslongas the cell
has alreadybeenmeasuredbythe UserEquipment(UE),whichisatypical scenario[8].Forthe purposes
of determininghandoverinterruptiontime,interactionswiththe core network(i.e.,networkentities
beyondthe radioaccessnetwork) are assumedtooccur in zerotime.Itis alsoassumedthatall

3. necessaryattributesof the targetchannel (thatis,downlinksynchronizationisachievedanduplink
access procedures,if applicable,are successfullycompleted) are knownatinitiationof the handover
fromthe servingchannel tothe targetchannel [9].ForanalysisRACHand PUCCH cycle istakenas 1ms.
The RACH and PUCCH waitingtimesinTDDcases are calculatedbasedonthe UL/DL sub-frame locations
inthe respective frame configurations.InTDDmode analysis,frame configuration1isconsidered.III.
ANALYSISThe intra- andinter-frequencyhandoverinterruptiontime iscalculatedbasedonthe
handoverprocedure showninfigure 3.The stepsinvolvedinhandoverinterruptionare:1) Radio
Synchronizationtothe targetcell.2) Average delaydue toRandomAccessCHannel (RACH) scheduling
period.3) RACH Preamble Transmission.4) Preamble detectionatTargeteNodeB.5) Transmissionof
RandomAccess(RA) - Time betweenthe RA responsetransmissionandUE’s receptionof scheduling
grant. 6) Decodingof schedulinggrantandtimingalignmentatUE. 7) Transmissionof data.Radio
synchronizationdelayisthe sumof the delaycausedbyfrequencysynchronizationanddownlink
synchronization.Frequencysynchronizationdelaydependsonwhetherthe targetcell isoperatingon
the same carrier frequencyasthe servingcell.Butsince the UE has alreadyidentifiedand
When the phone at the opposite end of the connection receives the RTP packets, it must
reassemble them back into an audio signal. If packets are missing, the audio signal will contain
gaps. This packet loss can be caused by a number of network problems. One
common cause of packet loss is congested WAN links.
There are two types of EPS bearers: default and dedicated. In the LTE network, the
EPS bearer QoS is controlled using the following LTE QoS parameters:
▶ Resource Type: GBR or Non-GBR
▶ QoS Parameters
 QCI
 ARP
 GBR
 MBR
 APN-AMBR
 UE-AMBR
Every EPS bearer must have QI and ARP defined. The QCI is particularly important
because it serves as reference in determining QoS level for each EPS bearer. In case
of bandwidth (bit rate), GBR and MBR are defined only in GBR type EPS bearers,
whereas AMBR (APN-AMBR and UE-AMBR) is defined only in Non-GBR type EPS
bearers.
Below, we will explain the LTE QoS parameters one by one.
Resource Type = GBR (Guaranteed Bit Rate)
For an EPS bearer, having a GBR resource type means the bandwidth of the bearer is

4. guaranteed. Obviously, a GBR type EPS bearer has a "guaranteed bit rate" associated
(GBR will be further explained below) as one of its QoS parameters. Only a dedicated
EPS bearer can be a GBR type bearer and no default EPS bearer can be GBR type. The
QCI of a GBR type EPS bearer can range from 1 to 4.
Resource Type = Non-GBR
For an EPS bearer, having a non-GBR resource type means that the bearer is a best
effort type bearer and its bandwidth is not guaranteed. A default EPS bearer is always
a Non-GBR bearer, whereas a dedicated EPS bearer can be either GBR or non-GBR.
The QCI of a non-GBR type EPS bearer can range from 5 to 9.
QCI (QoS Class Identifier)
QCI, in an integer from 1 to 9, indicates nine different QoS performance
characteristics of each IP packet. QCI values are standardized to reference specific
QoS characteristics, and each QCI contains standardized performance characteristics
(values), such as resource type (GBR or non-GBR), priority (1~9), Packet Delay
Budget (allowed packet delay shown in values ranging from 50 ms to 300 ms), Packet
Error Loss Rate (allowed packet loss shown in values from 10-2 to 10-6. For more
specific values, search on Google for "3GPP TS 23.203" and see Table 6.1.7 in the
document. For example, QCI 1 and 9 are defined as follows:
QCI = 1
: Resource Type = GBR, Priority = 2, Packet Delay Budget = 100ms, Packet Error
Loss Rate = 10-2 , Example Service = Voice
QCI = 9
: Resource Type = Non-GBR, Priority = 9, Packet Delay Budget = 300ms, Packet Error
Loss Rate = 10-6, Example Service = Internet
QoS to be guaranteed for an EPS bearer or SDF varies depending on the QCI values
specified.
QCI, though a single integer, represents node-specific parameters that give the details
of how an LTE node handles packet forwarding (e.g. scheduling weights, admission
thresholds, queue thresholds, link layer protocol configuration, etc). Network
operators have their LTE nodes pre-configured to handle packet forwarding according
to the QCI value.
By pre-defining the performance characteristics of each QCI value and having them
standardized, the network operators can ensure the same minimum level QoS
required by the LTE standards is provided to different services/applications used in an
LTE network consisting of various nodes from multi-vendors.
QCI values seem to be mostly used by eNBs in controlling the priority of packets
delivered over radio links. That's because practically it is not easy for S-GW or P-GW,
in a wired link, to process packets and also forward them based on the QCI
characteristics all at the same time (As you may know, a Cisco or Juniper router would
not care about delay or error loss rate when it processes QoS of packets. It would
merely decide which packet to send first through scheduling (WFQ, DWRR, SPQ, etc.)
based on the priority of the packets (802.1p/DSCP/MPLS EXP)).
ARP (Allocation and Retention Priority)

5. When a new EPS bearer is needed in an LTE network with insufficient resources, an
LTE entity (e.g. P-GW, S-GW or eNB) decides, based on ARP (an integer ranging from
1 to 15, with 1 being the highest level of priority), whether to:
 remove the existing EPS bearer and create a new one (e.g. removing an EPS bearer
with low priority ARP to create one with high priority ARP); or
 refuse to create a new one.
So, the ARP is considered only when deciding whether to create a new EPS bearer or
not. Once a new bearer is created and packets are delivered through it, the ARP does
not affect the priority of the delivered packet, and thus the network node/entity
forwards the packets regardless of their ARP values.
One of the most representative examples of using the ARP is an emergency VoIP call.
So, an existing EPS bearer can be removed if a new one is required for a emergency
119 (911 in US, 112 in EC, etc) VoIP call.
GBR (UL/DL)
This parameter is used for a GBR type bearer, and indicates the bandwidth (bit rate)
to be guaranteed by the LTE network. It is not applied to a non-GBR bearer with no
guaranteed bandwidth (UL is for uplink traffic and DL is for downlink traffic).
MBR (UL/DL)
MBR is used for a GBR type bearer, and indicates the maximum bit rate allowed in the
LTE network. Any packets arriving at the bearer after the specified MBR is exceeded
will be discarded.
APN-AMBR (UL/DL)
As you read the foregoing paragraph, you may wonder why a non-GBR type bearer
does not have a "bandwidth limit"? In case of non-GBR bearers, it is the total
bandwidth of all the non-GBR EPS bearers in a PDN that is limited, not the individual
bandwidth of each bearer. And this restriction is controlled by APN-AMBR (UL/DL). As
seen in the figure above, there are two non-GBR EPS bearers, and their maximum
bandwidths are specified by the APN-AMBR (UL/DL). This parameter is applied at UE
(for UL traffic only) and P-GW (for both DL and UL traffic).
UE-AMBR (UL/DL)
In the figure above, APN-AMBR and UE-AMBR look the same. But, please take a look
at the one below.
A UE can be connected to more than one PDN (e.g. PDN 1 for Internet, PDN 2 for VoIP
using IMS, etc.) and it has one unique IP address for each of its all PDN connections.
Here, UE-AMBR (UL/DL) indicates the maximum bandwidth allowed for all the non-
GBR EPS bearers associated to the UE no matter how many PDN connections the UE
has. Other PDNs are connected through other P-GWs, this parameter is applied by
eNBs only.
How SRVCC works

6. The SRVCC implementation controls the transfer of calls in both directions.
LTE to legacy network handover
Handover from LTE to the legacy network is required when the user moves out of the LTE coverage
area. Using SRVCC, the handover is undertaken in two stages.
 Radio Access Technology transfer: The handover for the radio access network and this
is a well-established protocol that is in use for transfers from 3G to 2G for example.
 Session transfer: The session transfer is the new element that is required for SRVCC. It is
required to move the access control and voice media anchoring from the Evolved Packet
Core, EPC of the packet switched LTE network to the legacy circuit switched network.
During the handover process the CSCF within the IMS architecture maintains the control of the
whole operation.
Voice handover using SRVCC on LTE
The SRVCC handover process takes place in a number of steps:
1. The handover process is initiated by a request for session transfer from the IMS CSCF.
2. The IMS CSCF responds simultaneously with two commands, one to the LTE network, and
the other to the legacy network.
3. the LTE network receives a radio Access Network handover execution command through the
MME and LTE RAN. This instructs the user device to prepare to move to a circuit switched
network for the voice call.
4. The destination legacy circuit switched network receives a session transfer response
preparing it to accept the call from the LTE network.
5. After all the commands have been executed and acknowledged the call is switched to the
legacy network with the IMS CSCF still in control of the call.
Legacy network to LTE
When returning a call to the LTE network much of the same functionality is again used.
To ensure the VoLTE device is able to return to the LTE RAN from the legacy RAN, there are two
options the legacy RAN can implement to provide a swift and effective return:

7.  Allow LTE information to be broadcast on the legacy RAN so the LTE device is able to
perform the cell reselection more easily.
 Simultaneously release the connection to the user device and redirect it to the LTE RAN.
SRVCC interruption performance
One of the key issues with VoLTE and SRVCC is the interruption time when handing over from an
LTE RAN to a legacy RAN.
The key methodology behind reducing he time is to simultaneous perform the redirections of RAN
and session. In this way the user experience is maintained and the actual interruption time is not
unduly noticeable.
It has been found that the session redirection is the faster of the two handovers, and therefore it is
necessary for the overall handover methodology to accommodate the fact that there are difference
between the two.