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Showing posts with label LTE Voice and SMS Issues. Show all posts
Showing posts with label LTE Voice and SMS Issues. Show all posts

Thursday, 13 February 2014

VoLTE Roaming with RAVEL (Roaming Architecture for Voice over IMS with Local Breakout)


Voice over LTE or VoLTE has many problems to solve. One of the issues that did not have a clear solution initially was Roaming. iBasis has a whitepaper on this topic here, from which the above picture is taken. The following is what is said above:

The routing of international calls has always been a problem for mobile operators. All too often the answer—particularly in the case of ‘tromboning’ calls all the way back to the home network—has been inelegant and costly. LTE data sessions can be broken out locally, negating the need for convoluted routing solutions. But in a VoIMS environment all of the intelligence that decides how to route the call resides in the home network, meaning that the call still has to be routed back.

The industry’s solution to this issue is Roaming Architecture for Voice over LTE with Local Breakout (RAVEL). Currently in the midst of standardisation at 3GPP, RAVEL is intended to enable the home network to decide, where appropriate, for the VoIMS call to be broken out locally. 

Three quarters of respondents to the survey said they support an industry-wide move to RAVEL for VoLTE roaming. This is emphatic in its enthusiasm but 25 per cent remains a significant share of respondents still to be convinced. Just over half of respondents said they plan to support VoIMS for LTE roaming using the RAVEL architecture, while 12.3 per cent said they would support it, but not using RAVEL.

Until RAVEL is available, 27.4 per cent of respondents said they plan to use home-routing for all VoLTE traffic, while just under one fifth said they would use a non-standard VoLTE roaming solution.

Well, the solution was standardised in 3GPP Release-11. NTT Docomo has an excellent whitepaper (embedded below) explaining the issue and the proposed solution.

In 3GPP Release 11, the VoLTE roaming and interconnection architecture was standardized in cooperation with the GSMA Association. The new architecture is able to implement voice call charging in the same way as circuit-switched voice roaming and interconnection models by routing both C-Plane messages and voice data on the same path. This was not possible with the earlier VoLTE roaming and interconnection architecture.

Anyway, here is the complete whitepaper




Monday, 20 January 2014

Different flavours of SRVCC (Single Radio Voice Call Continuity)



Single Radio Voice Call Continuity (SRVCC) has been quietly evolving with the different 3GPP releases. Here is a quick summary of these different flavors

In its simplest form, SRVCC comes into picture when an IMS based VoLTE call is handed over to the existing 2G/3G network as a normal CS call. SRVCC is particularly important when LTE is rolled out in small islands and the operator decided to provide VoLTE based call when in LTE. An alternative (used widely in practice) is to use CS Fallback (CSFB) as the voice option until LTE is rolled out in a wider area. The main problem with CSFB is that the data rates would drop to the 2G/3G rates when the UE falls back to the 2G/3G network during the voice call.



The book "LTE-Advanced: A Practical Systems Approach to Understanding 3GPP LTE Releases 10 and 11 Radio Access Technologies" by Sassan Ahmadi has some detailed information on SRVCC, the following is an edited version from the book:

SRVCC is built on the IMS centralized services (ICS) framework for delivering voice and messaging services to the users regardless of the type of network to which they are attached, and for maintaining service continuity for moving terminals.

To support GSM and UMTS, some modifications in the MSC server are required. When the E-UTRAN selects a target cell for SRVCC handover, it needs to indicate to the MME that this handover procedure requires SRVCC. Upon receiving the handover request, the MME triggers the SRVCC procedure with the MSC server. The MSC then initiates the session transfer procedure to IMS and coordinates it with the circuit-switched handover procedure to the target cell.

Handling of any non-voice packet-switched bearer is by the packet-switched bearer splitting function in the MME. The handover of non-voice packet-switched bearers, if performed, is according to a regular inter-RAT packet-switched handover procedure.

When SRVCC is enacted, the downlink flow of voice packets is switched toward the target circuit-switched network. The call is moved from the packet-switched to the circuit-switched domain, and the UE switches from VoIP to circuit-switched voice.

3GPP Rel-10 architecture has been recommended by GSMA for SRVCC because it reduces both voice interruption time during handover and the dropped call rate compared to earlier configurations. The network controls and moves the UE from E-UTRAN to UTRAN/GERAN as the user moves out of the LTE network coverage area. The SRVCC handover mechanism is entirely network-controlled and calls remain under the control of the IMS core network, which maintains access to subscribed services implemented in the IMS service engine throughout the handover process. 3GPP Rel-10 configuration includes all components needed to manage the time-critical signaling between the user’s device and the network, and between network elements within the serving network, including visited networks during roaming. As a result, signaling follows the shortest possible path and is as robust as possible, minimizing voice interruption time caused by switching from the packet-switched core network to the circuit-switched core network, whether the UE is in its home network or roaming. With the industry aligned around the 3GPP standard and GSMA recommendations, SRVCC-enabled user devices and networks will be interoperable, ensuring that solutions work in many scenarios of interest.

Along with the introduction of the LTE radio access network, 3GPP also standardized SRVCC in Rel-8 specifications to provide seamless service continuity when a UE performs a handover from the E-UTRAN to UTRAN/GERAN. With SRVCC, calls are anchored in the IMS network while the UE is capable of transmitting/ receiving on only one of those access networks at a given time, where a call anchored in the IMS core can continue in UMTS/GSM networks and outside of the LTE coverage area. Since its introduction in Rel-8, the SRVCC has evolved with each new release, a brief summary of SRVCC capability and enhancements are noted below

3GPP Rel-8: Introduces SRVCC for voice calls that are anchored in the IMS core network from E-UTRAN to CDMA2000 and from E-UTRAN/UTRAN (HSPA) to UTRAN/GERAN circuit-switched. To support this functionality, 3GPP introduced new protocol interface and procedures between MME and MSC for SRVCC from E-UTRAN to UTRAN/GERAN, between SGSN and MSC for SRVCC from UTRAN (HSPA) to UTRAN/GERAN, and between the MME and a 3GPP2-defined interworking function for SRVCC from E-UTRAN to CDMA 2000.

3GPP Rel-9: Introduces the SRVCC support for emergency calls that are anchored in the IMS core network. IMS emergency calls, placed via LTE access, need to continue when SRVCC handover occurs from the LTE network to GSM/UMTS/CDMA2000 networks. This evolution resolves a key regulatory exception. This enhancement supports IMS emergency call continuity from E-UTRAN to CDMA2000 and from E-UTRAN/UTRAN (HSPA) to UTRAN/ GERAN circuit-switched network. Functional and interface evolution of EPS entities were needed to support IMS emergency calls with SRVCC.

3GPP Rel-10: Introduces procedures of enhanced SRVCC including support of mid-call feature during SRVCC handover (eSRVCC); support of SRVCC packet-switched to circuit-switched transfer of a call in alerting phase (aSRVCC); MSC server-assisted mid-call feature enables packet-switched/ circuit-switched access transfer for the UEs not using IMS centralized service capabilities, while preserving the provision of mid-call services (inactive sessions or sessions using the conference service). The SRVCC in alerting phase feature adds the ability to perform access transfer of media of an instant message session in packet-switched to circuit-switched direction in alerting phase for access transfers.

3GPP Rel-11: Introduces two new capabilities: single radio video call continuity for 3G-circuit-switched network (vSRVCC); and SRVCC from UTRAN/GERAN to E-UTRAN/HSPA (rSRVCC). The vSRVCC feature provides support of video call handover from E-UTRAN to UTRAN-circuitswitched network for service continuity when the video call is anchored in IMS and the UE is capable of transmitting/receiving on only one of those access networks at a given time. Service continuity from UTRAN/GERAN circuitswitched access to E-UTRAN/HSPA was not specified in 3GPP Rel-8/9/10. To overcome this drawback, 3GPP Rel-11 provided support of voice call continuity from UTRAN/GERAN to E-UTRAN/HSPA. To enable video call transfer from E-UTRAN to UTRAN-circuit-switched network, IMS/EPC is evolved to pass relevant information to the EPC side and S5/S11/Sv/Gx/Gxx interfaces are enhanced for video bearer-related information transfer. To support SRVCC from GERAN to E-UTRAN/HSPA, GERAN specifications are evolved to enable a mobile station and base station sub-system to support seamless service continuity when a mobile station hands over from GERAN circuit-switched access to EUTRAN/ HSPA for a voice call. To support SRVCC from UTRAN to EUTRAN/ HSPA, UTRAN specifications are evolved to enable the RNC to perform rSRVCC handover and to provide relative UE capability information to the RNC.

NTT Docomo has a presentation on SRVCC and eSRVCC which is embedded below:



Saturday, 31 August 2013

VoLTE Bearers

While going through Anritsu whitepaper on VoLTE, I found this picture that explains the concepts of bearers in a VoLTE call well. From the whitepaper:

All networks and mobile devices are required to utilize a common access point name (APN) for VoLTE, namely, “IMS”. Unlike many legacy networks, LTE networks employ the “always-on” conception of packet connectivity: Devices have PDN connectivity virtually from the moment they perform their initial attach to the core network. During the initial attach procedure, some devices choose to name the access point through which they prefer to connect. However, mobile devices are not permitted to name the VoLTE APN during initial attach, i.e., to utilize the IMS as their main PDN, but rather to establish a connection with the IMS AP separately. Thus, VoLTE devices must support multiple simultaneous default EPS bearers.

Note that because the VoLTE APN is universal, mobile devices will always connect through the visited PLMN’s IMS PDN-GW. This architecture also implies the non-optionality of the P-CSCF:

As stated, VoLTE sessions employ two or three DRBs. This, in turn, implies the use of one default EPS bearer plus one or two dedicated EPS bearers. The default EPS bearer is always used for SIP signaling and exactly one dedicated EPS bearer is used for voice packets (regardless of the number of active voice media streams.) XCAP signaling may be transported on its own dedicated EPS bearer – for a total of three active EPS bearers – or it may be multiplexed with the SIP signaling on the default EPS bearer, in which case only two EPS bearers are utilized.

My understanding is that initially when the UE is switched on, a default bearer with QCI 9 (see old posts on QoS/QCI here) is established that would be used for all the signalling. Later on, another default bearer with QCI 5 is established with the IMS CN. When a VoLTE call is being setup, a dedicated bearer with QCI 1 is setup for the voice call. As the article says, another dedicated bearer may be needed for XCAP signalling. If a Video call on top of VoLTE is being used than an additional dedicated bearer with QCI 2 will be setup. Note that the voice pat will still be carried by dedicated bearer with QCI 1.

Do you disagree or have more insight, please feel free to add the comment at the end of the post.

The whitepaper is embedded below and is available to download from slideshare.



Related posts:

Tuesday, 8 January 2013

VoLTE, Battery Issues and Solutions


Sometime back we had news about how VoLTE is battery killer and how it would suck our 4G phones dry. Well, I agree. I am no fan of VoLTE and think that CSFB solution can suffice in mid-term. Having said that, there is a solution which would be soon available to sort this battery issue during VoLTE call. I had a post on this topic earlier titled SPS and TTI Bundling. I am not sure about exactly how much saving would occur if either of the features are implemented.

ST Ericsson has recently released a whitepaper on this topic that is embedded below. If you have more idea on this, please add it in comments.



Wednesday, 7 November 2012

CSFB Performance

Here is another presentation from Qualcomm from the '4G World'.



With regards to SI Tunneling mentioned in the presentation, I found the following in another Qualcomm whitepapers:


With Release 9 Enhanced Release with Redirection—SI Tunneling, the device follows 3GPP release 9, where SIB information can be tunneled from the target Radio Access Network (RAN) via the core network to the source RAN and be included in the redirection message sent to the device. This can avoid reading any SIBs on the target cell. 

The predominant solutions deployed today are based on Release 8 Release with Redirection — SIB Skipping, in order to achieve good call setup times, good reliability, and simplify deployments. It is anticipated that Release 9 Enhanced Release with Redirection will be deployed in the near future. At this time, there is not as much push for handover-based CSFB since both Release 8 Release with Redirection—SIB Skipping and Release 9 Enhanced Release with Redirection—SI Tunneling have largely addressed any call setup time issues that may have existed with the Basic Release with Redirection solution.


I have blogged on this topic before, here.

More on Dual Radio here and SVLTE here.

Tuesday, 6 November 2012

17 LTE Voice Modes

No wonder why LTE chipsets are complicated.


From Qualcomm's presentation in 4G World, available here.

Sunday, 26 August 2012

Voice-Over-LTE (VoLTE) Signalling

MetroPCS has recently launched rolled out VoLTE in USA using LG connect phones. More operators would be rolling it out soon so here is example of Signaling in VoLTE.




To read in detail, please see the article from NTT Docomo technical journal here.

Saturday, 19 May 2012

SPS and TTI Bundling Example

I have blogged about Semi-Persistent Scheduling (SPS) and Transmit Time Interval (TTI) Bundling feature before. They are both very important for VoIP and VoLTE to reduce the signalling overhead.



It should be noted that as per RRC Specs, SPS and TTI Bundling is mutually exclusive. The following from RRC specs:

TTI bundling can be enabled for FDD and for TDD only for configurations 0, 1 and 6. For TDD, E-UTRAN does not simultaneously enable TTI bundling and semi-persistent scheduling in this release of specification. Furthermore, E-UTRAN does not simultaneously configure TTI bundling and SCells with configured uplink.

Wednesday, 16 May 2012

Thursday, 23 February 2012

High level view on how SMS works in LTE


The following is from E\\\ whitepaper available here:


In 2010, 6.9 trillion text messages were sent globally and this figure is expected to break the eight trillion mark in 2011. This represents USD 127 billion in revenue for operators. LTE provides the same basic SMS features, such as concatenated SMS, delivery notification and configuration. However, the SMS delivery mechanism is somewhat different. A VoLTE device can send and receive text messages encapsulated within a SIP message. To receive a text message, the encapsulation process is invoked by an IP short-message-gateway in the IMS domain, and the gateway converts traditional Signaling System Number 7 (SS7) Mobile Application Part (MAP) signaling to IP/SIP.


To ensure that text messages are routed via the gateway, the home location register (HLR) of the recipient needs an additional function to return a routable gateway address back to the SMS-C on receipt of an SMS-routing request.


When a VoLTE device sends a text message, it should perform the encapsulation. The gateway extracts the text message inside a SIP MESSAGE signal before passing it on to the SMS-C.


However, if the VoLTE device is configured to not invoke SMS over IP networks, text messages can be sent and received over LTE without the need for any SIP encapsulation. A received text message will reach the mobile switching center server (MSC-S) of the mobile softswitch system in the same way as it does today. The MSC-S will page the device via the SGs interface with the Mobile Management Entity (MME) of the EPC system. Once a paging response is received, the MSC-S will pass the SMS on to the MME, which in turn tunnels it onto the device. Due to the support for SMS delivery and IP connectivity provided by LTE/EPC, MMS works seamlessly.


For more technically minded people, there is a whitepaper that covers SMS in detail available here.

Wednesday, 30 November 2011

Reducing CSFB Timing with RRC R9 Optimisations

While in the initial testing CSFB timing used to be between 6-8 seconds, most Rel-8 phones can complete the CSFB procedure between 4-4.5 seconds. Unfortunately this is still a lot in terms of signalling. To reduce this in Rel-9 there is a simple optimisation that has been done.
In the RRC Connection Release message, there is a possibility to add UTRAN/GERAN System Information messages. In the picture above, I have only shown UTRA System Information but a similar picture can be drawn for GERAN.

Once all the Mandatory SIB's are sent to the UE then it can immediately camp on without the need to read any other additional system info. This will reduce the CSFB time between 1-2 seconds.

The lesser the CSFB time, the better the Quality of end user experience

Tuesday, 18 October 2011

HD Voice - Next step in the evolution of voice communication

Nearly 2 years back I blogged about Orange launching HD Voice via the use of AMR-WB (wideband) codecs. HD voice is already fully developed and standardized technology and has so far been deployed on 32 networks in almost as many countries.

People who have experienced HD voice say it feels like they are talking to a person in the same room. Operators derive 70 percent of their revenue from voice and voice-related services, and studies show that subscribers appreciate the personal nature of voice communication, saying it offers a familiar and emotional connection to another person.

HD voice is also a reaction to the competition faced by the operators from OTT players like Skype.

Below is an embed from the recent whitepaper by Ericsson:

For more information also see:



Wednesday, 5 October 2011

Simultaneous Voice and LTE (SVLTE)


When LTE is an overlay to a CDMA/EV-DO network, the current de facto standard for voice delivery is Simultaneous Voice and LTE (SVLTE). In this arrangement, voice service is deployed as a 1x service running in parallel with LTE data services. For this solution to work, the handset needs to have two radios that are on simultaneously. The problem that is obvious is that the power consumption would generally be higher as two radios are on when the voice call is ongoing. The advantage (and I think its a big advantage) is that the data speeds are not affected by ongoing voice call and at the same time the state machine is simple.

For some reason this idea is not very popular for the 2G/3G evolution to LTE as the reliance will be on the CS Fallback. I had discussed this idea in the LTE World Summit and had blogged about it, you can read more details and comments here.

There is also a recent whitepaper from Huawei that covers these issues going towards VoLTE. Its available here.

Edit 06/10/11: Changed the acronym of SVLTE from 'Simultaneous Voice Over LTE' to 'Simultaneous Voice and LTE' as this is correct and referred to elsewhere.

Friday, 27 May 2011

Dual Radio Solution for Voice in LTE

I did mention in the Twitter conversations post from LTE World Summit 2011 that there are now certain analysts and players in the market who think that it should be possible to have two radios. Here is a slide from ZTE that shows that they are thinking in this direction as well.


Click on the pic to enlarge.

Tri-SIM phones have been available for quite a while but now there are Quad-Sim Shanzhai phones that are available in China. I am sure there is a market for these kind of phones.

With the battery life and the mobile technology improving, these are no longer the concerns when talking about dual radio possibility in the devices. Another common argument is that there may be additional interference due to multiple radios simultaneously receiving/transmitting. I am sure these can be managed without much problem.

Another problem mentioned is we may need multiple SIM cards but the SIM cards used is actually a UICC. There can be multiple SIM applications and IMSI's on it. The network may need some very minor modifications but they should be able to manage this with no problems. In the good old days, we used to have mobiles with built in Fax. The mobile number used to be different from the Fax number. It was a similar kind of problem but managed without problem.

So there may still be time to keep LTE simple by standardising the dual-radio solution rather than having CSFB, VoLTE, SRVCC, VoLGA, etc.

Any thoughts?

Monday, 25 April 2011

Advanced Telephony Services for LTE

With LTE World Summit just round the corner, I was going through the last year's presentations and realised that we didn't talk of this one before.

The concept for the advanced telephony services has been around since the early days of IMS and this was one of the ways IMS was sold. Unfortunately IMS didn't take off as planned and only now with the standardisation of VoLTE, there is a possibility of the advanced services becoming a reality.

The following presentation summarises some of these advanced telephony services concepts.

Wednesday, 23 February 2011

Circuit Switched Fallback (CSFB): A Quick Primer

I have explained CSFB with basic signalling here and there is a very interesting Ericsson whitepaper explaining all Voice issues in LTE here.

The following CSFB details have been taken from NTT Docomo Technical Journal:

The basic concept of CS Fallback is shown in Figure 1. Given a mobile terminal camping on LTE, a mobile terminating voice call arrives at the terminal from the existing CS domain via EPC. On receiving a paging message, the mobile terminal recognises that the network is calling the mobile terminal for CS-based voice and therefore switches to 3G. The response confirming the acceptance of a call request is then sent from the mobile terminal to the 3G-CS system, and from that point on, all call control for the voice service is performed on the 3G side.

The CS Fallback consists of a function to notify a mobile terminal of a call request from the CS domain and combined mobility management functions between CS domain and EPC for that
purpose. The network architecture of CS Fallback is shown in Figure 2.


One of the remarkable characteristics of the EPC supporting CS Fallback is that it connects the Mobile Switching Center (MSC) and Visited Location Register (VLR) in the 3G CS domain
with the Mobility Management Entity (MME), which provides EPC mobility management functionality. The interface connecting MSC/VLR and MME is called an SGs reference point. This
interface is based on the concept of the Gs reference point that exchanges signalling with MSC, which connects to the Serving General Packet Radio Service Support Node (SGSN), a 3G
packet switch. The SGs provides nearly all the functions provided by the existing Gs.

The CS Fallback function uses this SGs reference point to transfer the mobile terminating call requests from the CS domain to LTE. It also provides combined mobility management
between the 3G CS domain and the EPC to enable this transfer to take place.


Combined Mobility Management between CS Domain and EPC Network:

A mobile communications network must always know where a mobile terminal is located to deliver mobile terminating service requests to the mobile user on the mobile terminating side. The procedure for determining terminal location is called “mobility management". As a basic function of mobile communications, 3G and LTE each provide a mobility management function.

To complete a call using the CS Fallback function, the CS domain needs to know which LTE location registration area the mobile terminal is currently camping on. To this end, the MME must correlate mobility management control of the CS domain with that of EPC and inform MSC/VLR that the mobile terminal is present in an LTE location registration area.

The 3G core network already incorporates a function for linking mobility management of the CS domain with that of the Packet Switched (PS) domain providing packet-switching functions. As described above, the CS domain and PS domain functions are provided via separate switches. Thus, if combined mobility management can be used, the mobility management procedure for the terminal only needs to be performed once, which has the effect of reducing signal traffic in the network. This concept of combined mobility management is appropriated by the CS Fallback function. Specifically, MSC/VLR uses the same logic for receiving a location registration request from SGSN as that for receiving a location registration request from MME. This achieves a more efficient combined mobility management between the CS domain and EPC while reducing the development impact on MSC.

As described above, a mobile terminal using LTE cannot use 3G at the same time. This implies that the MME, which contains the LTE location registration area (Tracking Area (TA)), is unable to identify which MSC/VLR it should send the mobility management messages to from the TA alone. To solve this problem, the mapping of TAs and 3G Location Areas (LA) within MME has been adopted. The concept behind TA/LA mapping is shown in Figure 3. Here, MME stores a database that manages the correspondence between physically overlapping TAs and LAs. This information is used to determine which MSC/VLR to target for location registration.

The combined TA/LA update procedure for CS fallback is shown in detail in Figure 4. First, the mobile terminal sends to the MME a Tracking Area Update (TAU) request message indicating a combined TAU and the current TA in which the mobile terminal is currently present (Fig. 4 (1)). The MME then performs a location update procedure towards Home Subscriber Server (HSS), which is a database used for managing subscriber profiles (Fig. 4 (2)). Next, the MME uses the TA/LA correspondence database to identify the corresponding LA and the MSC/VLR that is managing that area, and uses the SGs reference point to send a Location Area Update (LAU) request message to the MSC/VLR together with the LA so identified (Fig. 4 (3)). The MSC/VLR that receives the LAU request message stores the correspondence between the ID of the MME originating the request and an ID such as the International Mobile Subscriber Identity (IMSI) that identifies the subscriber (Fig. 4 (4)). This enables the MSC/VLR to know which MME the mobile terminal is currently connected to and that the mobile terminal is camping on LTE. Following this, the MSC/VLR performs a location registration procedure with the HSS (Fig. 4 (5)). Finally, the MSC/VLR informs the MME of temporary user identity (Temporary Mobile Subscriber Identity (TMSI)), which is used at the time of a mobile terminating call in the CS domain, and indicates that location registration has been completed. The MME then informs the mobile terminal of the TMSI and of the LA that the mobile terminal has been registered with thereby completing combined location registration (Fig. 4 (6) (7)).

CS Fallback Call Control Procedures - Mobile Originating Call:


To originate a voice call using the CS Fallback function, a mobile terminal in the LTE location registration area must first switch (fall back) to 3G. The mobile-originating voice call procedure is shown in Figure 5. To originate a call, the mobile terminal begins by sending a CS fallback service request message to the MME (Fig. 5 (1)). Since a packet-communications transmission path (bearer) must always exist in EPC for the purpose of providing an always-on connection, the bearer also has to be handed over to 3G. To accomplish this, the MME issues a handover command to the mobile terminal in LTE and initiates a handover procedure (Fig. 5 (2)). The mobile terminal changes its radio from LTE to 3G during this procedure (Fig. 5 (3)). On completion of handover, the mobile terminal issues an originating request for voice service to the MSC/VLR. A voice-call connection is then established using an existing calloriginating procedure on 3G and the CS Fallback procedure is completed (Fig. 5(4)).

CS Fallback Call Control Procedures - Mobile Terminating Call:

The mobile terminating voice call procedure using CS Fallback is shown in Figure 6. When the MSC/VLR receives a message indicating the occurrence of a mobile terminating call (Fig. 6 (1)), the MSC/VLR identifies the corresponding MME from the call information received (Fig. 6 (2)). Then, the MSC/VLR sends a paging message (Fig. 6 (3)) towards the MME. Next, the MME sends a paging message to the mobile terminal in LTE (Fig. 6 (4)). This paging message includes an indication that the call is a CS service, and on identifying the call as such, the mobile terminal sends a CS fallback service request signal to the MME (Fig. 6 (5)). Following this, a handover procedure to 3G as described above takes place (Fig. 6 (6), (7)). The mobile terminal that is now switched to 3G sends a paging response message to the MSC/VLR at which it is registered (Fig. 6 (8)). Finally, an existing mobile terminating call procedure on 3G is executed and the CS Fallback procedure is completed (Fig. 6 (9)).

Tuesday, 8 February 2011

VoLTE: Semi-Persistent Scheduling (SPS) and TTI Bundling

The following is from the recently released 4G Americas paper '4G Mobile Broadband Evolution: 3GPP Release-10 and beyond:

With the support of emergency and location services in Rel-9, interest in Voice over LTE (VoLTE) has increased. This is because the Rel-9 enhancements to support e911 were the last step to enable VoLTE (at least in countries that mandate e911) since the Rel-8 specifications already included the key LTE features required to support good coverage, high capacity/quality VoLTE. There are two main features in Rel-8 that focus on the coverage, capacity and quality of VoLTE: Semi-Persistent Scheduling (SPS) and TTI Bundling.

SPS is a feature that significantly reduces control channel overhead for applications that require persistent radio resource allocations such as VoIP. In LTE, both the DL and UL are fully scheduled since the DL and UL traffic channels are dynamically shared channels. This means that the physical DL control channel (PDCCH) must provide access grant information to indicate which users should decode the physical DL shared channel (PDSCH) in each subframe and which users are allowed to transmit on the physical UL shared channel (PUSCH) in each subframe. Without SPS, every DL or UL physical resource block (PRB) allocation must be granted via an access grant message on the PDCCH. This is sufficient for most bursty best effort types of applications which generally have large packet sizes and thus typically only a few users must be scheduled each subframe. However, for applications that require persistent allocations of small packets (i.e. VoIP), the access grant control channel overhead can be greatly reduced with SPS.

SPS therefore introduces a persistent PRB allocation that a user should expect on the DL or can transmit on the UL. There are many different ways in which SPS can setup persistent allocations, and Figure below shows one way appropriate for VoLTE. Note that speech codecs typically generate a speech packet every 20 ms. In LTE, the HARQ interlace time is 8 ms which means retransmissions of PRBs that have failed to be decoded can occur every 8 ms. Figure below shows an example where a maximum of five total transmissions (initial transmission plus four retransmissions) is assumed for each 20 ms speech packet with two parallel HARQ processes. This figure clearly shows that every 20 ms a new “first transmission” of a new speech packet is sent. This example does require an additional 20 ms of buffering in the receiver to allow for four retransmissions, but this is generally viewed as a good tradeoff to maximize capacity/coverage (compared to only sending a maximum of two retransmissions).

The example in Figure above can be applied to both the DL and UL and note that as long as there are speech packets arriving (i.e. a talk spurt) at the transmitter, the SPS PRBs would be dedicated to the user. Once speech packets stop arriving (i.e. silence period), these PRB resources can be re-assigned to other users. When the user begins talking again, a new SPS set of PRBs would be assigned for the duration of the new talkspurt. Note that dynamic scheduling of best effort data can occur on top of SPS, but the SPS allocations would take precedent over any scheduling conflicts.


TTI bundling is another feature in Rel-8 that optimizes the UL coverage for VoLTE. LTE defined 1 ms subframes as the Transmission Time Interval (TTI) which means scheduling occurs every 1 ms. Small TTIs are good for reducing round trip latency, but do introduce challenges for UL VoIP coverage. This is because on the UL, the maximum coverage is realized when a user sends a single PRB spanning 180 kHz of tones. By using a single 180 kHz wide PRB on the UL, the user transmit power/Hz is maximized. This is critical on the UL since the user transmit power is limited, so maximizing the power/Hz improves coverage. The issue is that since the HARQ interlace time is 8 ms, the subframe utilization is very low (1/8). In other words, 7/8 of the time the user is not transmitting. Therefore, users in poor coverage areas could be transmitting more power when a concept termed TTI bundling (explained in the next paragraph) is deployed.

While it’s true that one fix to the problem is to just initiate several parallel HARQ processes to fill in more of the 7/8 idle time, this approach adds significant IP overhead since each HARQ process requires its own IP header. Therefore, TTI bundling was introduced in Rel-8 which combined four subframes spanning 4 ms. This allowed for a single IP header over a bundled 4 ms TTI that greatly improved the subframe utilization (from 1/8 to 1/2) and thus the coverage (by more than 3 dB).

Martin Sauter puts it in a simpler way in his blog as follows: The purpose of TTI Bundling is to improve cell edge coverage and in-house reception for voice. When the base station detects that the mobile can't increase it's transmission power and reception is getting worse it can instruct the device to activate TTI bundling and send the same packet but with different error detection and correction bits in 2, 3 or even 4 consecutive transmit time intervals. The advantage over sending the packet in a single TTI and then detecting that it wasn't received correctly which in turn would lead to one or more retransmissions is that it saves a lot of signaling overhead. Latency is also reduced as no waiting time is required between the retransmissions. In case the bundle is not received correctly, it is repeated in the same way as an ordinary transmission of a packet. Holma and Toskala anticipate a 4dB cell edge gain for VoIP with this feature which is quite a lot. For details how the feature is implemented have a look at 3GPP TS 36.321.

A whitepaper explaining the concepts of TTI Bundling is available on Slideshare here.