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

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.

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.

Thursday, 20 January 2011

eMPS: Enhanced Multimedia Priority Service in Release-10 and beyond


The response to emergency situations (e.g., floods, hurricanes, earthquakes, terrorist attacks) depends on the communication capabilities of public networks. In most cases, emergency responders use private radio systems to aid in the logistics of providing critically needed restoration services. However, certain government and emergency management officials and other authorised users have to rely on public network services when the communication capability of the serving network may be impaired, for example due to congestion or partial network infrastructure outages, perhaps due to a direct or indirect result of the emergency situation.

Multimedia Priority Service, supported by the 3GPP system set of services and features, is one element creating the ability to deliver calls or complete sessions of a high priority nature from mobile to mobile networks, mobile to fixed networks, and fixed to mobile networks.

Requirements for the Multimedia Priority Service (MPS) have been specified in TS 22.153 for the 3GPP Release-9

The intention of MPS is to enable National Security/Emergency Preparedness (NS/EP) users (herein called Service Users) to make priority calls/sessions using the public networks during network congestion conditions. Service Users are the government-authorized personnel, emergency management officials and/or other authorized users. Effective disaster response and management rely on the Service User’s ability to communicate during congestion conditions. Service Users are expected to receive priority treatment, in support of mission critical multimedia communications.

LTE/EPC Release 9 supports IMS-based voice call origination by a Service User and voice call termination to a Service User with priority. However, mechanisms for completing a call with priority do not exist for call delivery to a regular user for a priority call originated by a Service User. MPS enhancements are needed to support priority treatment for Release 10 and beyond for call termination and for the support of packet data and multimedia services.

MPS will provide broadband IP-based multimedia services (IMS-based and non-IMS-based) over wireless networks in support of voice, video, and data services. Network support for MPS will require end-to-end priority treatment in call/session origination/termination including the Non Access Stratum (NAS) and Access Stratum (AS) signaling establishment procedures at originating/terminating network side as well as resource allocation in the core and radio networks for bearers. The MPS will also require end-to-end priority treatment in case of roaming if supported by the visiting network and if the roaming user is authorized to receive priority service.

MPS requirement is already achieved in the 3G circuit-switched network. Therefore, if the network supports CS Fallback, it is necessary to provide at least the same capability as 3G circuit switched-network in order not to degrade the level of voice service. In CS Fallback, UE initiates the fallback procedures over the LTE as specified in TS 23.272 when UE decides to use the CS voice service for mobile originating and mobile terminating calls. To achieve priority handling of CS Fallback, NAS and AS signaling establishment procedures, common for both IP-based multimedia services and CS Fallback, shall be treated in a prioritized way.

In Release-10, for LTE/EPC, the following mechanisms will be specified.
  • Mechanisms to allocate resources for signaling and media with priority based on subscribed priority or based on priority indicated by service signaling.
  • For a terminating IMS session over LTE, a mechanism for the network to detect priority of the session and treat it with priority.
In Release-10, for CS Fallback, the following mechanism will be specified:
  • A mechanism to properly handle the priority terminating voice call and enable the target UE to establish the AS and NAS connection to fall-back to the GERAN/UTRAN/1xRTT.
For more information, see:

3GPP TR 23.854: Enhancements for Multimedia Priority Service (Release 10)

3GPP TS 22.153: Multimedia priority service (Release 10)

Friday, 3 December 2010

Presentation: IMS for 3G Voice Services and Migration Strategies

Very interesting presentation from NTT DoCoMo in the IMS workshop I blogged about yesterday. It shows their strategy to move from legacy core network to an All IP Network (AIPN).


Saturday, 6 November 2010

LTE CS Fallback Procedure

From a 3GPP presentation by Hannu Hietalahti:

1. CS FallBack from EPS to CS domain
2. CSFB reuses voice and other CS-domain services provided by legacy CS infrastructure
3. EPS redirects the UE to CS Domain for CS services
3a. SMS can be delivered to the UE without redirecting to CS Domain
3b. After CS service the UE returns to LTE, depending on coverage and policy
4. User can decide, based on CLI, whether to accept CSFB request
5. Application of CSFB:
5a. CS capable device camping on LTE cell can establish/receive CS services
5b. Reuse of existing CS infrastructure for voice service until IMS VoIP is deployed
5c. Provide voice roaming support with LTE
5d. Support E911 using existing CS infrastructure
5e. Rel-9 IMS provides full emergency call support
5f. Requires overlapping CS domain coverage

Note: CSFB applies between LTE and GSM, WCDMA and 1xRTT