Showing posts with label Orange. Show all posts
Showing posts with label Orange. Show all posts

Friday, 22 October 2010

IMB and TDtv (and DVB-H)

Its long time since I blogged about TDtv. Its been quite a while since I heard about TDtv. Apparently its been superseded by IMB, aka. Integrated Mobile Broadcast.

IMB is used to stream live video and store popular content on the device for later consumption. This results in a significant offloading of data intensive traffic from existing 3G unicast networks and an improved customer experience. The multimedia client features an intuitive electronic program guide, channel grid and embedded video player for live TV viewing and video recording. All IMB applications can be quickly and cost-effectively adapted to support all major mobile operating systems and different mobile device types, including smartphones, tablets and e-readers.

IMB was defined in the 3GPP release 8 standards, and was recently endorsed by the GSMA as their preferred method for the efficient delivery of broadcast services. In June 2010, O2, Orange and Vodafone – three of the five major UK mobile operators – announced that they have teamed up for a three-month trial that will explore IMB wireless technology within a tranche of 3G TDD spectrum.

This spectrum already forms part of the 3G licenses held by many European mobile operators, but has remained largely unused because of a lack of appropriate technology. Currently, 3G TDD spectrum is available to over 150 operators across 60 countries, covering more than half a billion subscribers. IMB enables spectrally efficient delivery of broadcast services in the TDD spectrum based on techniques that are aligned with existing FDD WCDMA standards. This enables a smooth handover between IMB and existing 3G networks.

Issues that previously limited uptake of IMB, or IPWireless' tdTV system, have now all been addressed. Namely, the standard now allows for smooth handover between IMB and unicast delivery; has the potential to be integrated onto a single W-CDMA chip rather than requiring a separate chip; and has resolved interference issues with FDD W-CDMA, at least for spectrum in the 1900MHz to 1910MHz range.

IP Wireless already had a trial at Orange and T-Mobile in the UK (which have just agreed to merge), but in that pilot each 5MHz segment only gave rise to 14 TV channels per operator. The new standard could support 40 separate TV channels if two operators shared their TDD spectrum.

The GSMA announced its support and is backed up with additional support from both IPWireless and Ericsson as well as operators Orange, Softbank and Telstra.

There have been recently quite a few bad news for DVB-H and on top of that IP Wireless has announced that Samsung is going to be releasing phones with IMB support so it may be that we will see IMB sometime next year.

The GSMA paper that details IMB service scenarios and System requirements is embedded below:

Tuesday, 23 February 2010

Codec's for LTE

Sometime back I mentioned about Orange launching AMR-WB codec which would result in 'hi-fi quality' voice (even though its being referred to as HD voice by some). Since then, there has been not much progress on this HD-voice issue.

CODEC stands for “COder-DECoder,” but is also known as an enCOder-DECoder and COmpression-DECompression system when used in video systems. Codec's are important as they compress the voice/video data/packets so less bandwidth is required for the data to be transmitted. At the same time it has to be borne in mind that the capacity to withstand errors decrease with higher compression ratio and as a result it may be necessary to change the codecs during the voice/video call. This calls for flexibility as in case of AMR (Adaptive Multi Rate) Codecs.

The following is from Martin Sauter's book "Beyond 3G – Bringing Networks, Terminals and the Web Together":

Voice codecs on higher layers have been designed to cope with packet loss to a certain extent since there is not usually time to wait for a repetition of the data. This is why data from circuit-switched connections is not repeated when it is not received correctly but simply ignored. For IP sessions, doing the same is difficult, since a single session usually carries both real-time services such as voice calls and best-effort services such as Web browsing simultaneously. In UMTS evolution networks, mechanisms such as ‘Secondary PDP contexts’ can be used to separate the real-time data traffic from background or signaling traffic into different streams on the air interface while keeping a single IP address on the mobile device.

UMTS uses the same codecs as GSM. On the air interface users are separated by spreading codes and the resulting data rate is 30–60 kbit/s depending on the spreading factor. Unlike GSM, where timeslots are used for voice calls, voice capacity in UMTS depends less on the raw data rate but more on the amount of transmit power required for each voice call. Users close to the base station require less transmission power in downlink compared with more distant users. To calculate the number of voice calls per UMTS base station, an assumption has to be made about the distribution of users in the area covered by a cell and their reception conditions. In practice, a UMTS base station can carry 60–80 voice calls per sector. A typical three-sector UMTS base station can thus carry around 240 voice calls. As in the GSM example, a UMTS cell also carries data traffic, which reduces the number of simultaneous voice calls.

The following is an extract from 3G Americas white paper, "3GPP Mobile Broadband Innovation Path to 4G: Release 9, Release 10 and Beyond: HSPA+, SAE/LTE and LTE-Advanced,":

Real-time flows (voice/video) based on rate adaptive codecs can dynamically switch between different codec rates. Codec rate adaptation allows an operator to trade off voice/video quality on one side and network capacity (e.g. in terms of the number of accepted VoIP calls), and/or radio coverage on the other side. Operators have requested a standardized solution to control the codec rate adaptation for VoIP over LTE, and a solution has been agreed upon and specified in the 3GPP Rel-9 specifications, which is provided in this paper.


Given previous discussion in 3GPP (3GPP S4-070314) it was clear that dropping IP packets was not an acceptable means for the network to trigger a codec rate reduction. Instead an explicit feedback mechanism had to be agreed on by which the network (e.g. the eNodeB) could trigger a codec rate reduction. The mechanism agreed on for 3GPP Rel-9 is the IP-based Explicit Congestion Notification (ECN) specified in an IETF RFC. ECN is a 2 bit field in the end-to-end IP header. It is used as a “congestion pre-warning scheme” by which the network can warn the end points of incipient congestion so that the sending endpoint can decrease its sending rate before the network is forced to drop packets or excessive delay of media occurs. Any ECN-based scheme requires two parts: network behavior and endpoint behavior. The first part had already been fully specified in an IETF RFC106 and merely had to be adopted into the corresponding specifications (3GPP TS 23.401 and 3GPP TS 36.300). The network behavior is completely service and codec agnostic. That is, it works for both IMS and non-IMS based services and for any voice/video codec with rate-adaptation capabilities. The main work in 3GPP focused on the second part: the endpoint behavior. For 3GPP Rel-9, the endpoint behavior has been specified for the Multimedia Telephony Service for IMS (MTSI - 3GPP TS 26.114). It is based on a generic (i.e. non-service specific) behavior for RTP/UDP based endpoints, which is being standardized in the IETF.

Furthermore, it was agreed that no explicit feedback was needed from the network to trigger a codec rate increase. Instead, the Rel-9 solution is based on probing from the endpoints – more precisely the Initial Codec Mode (ICM) scheme that had already been specified in 3GPP Rel-7 (3GPP S4-070314). After the SIP session has been established, the sending side always starts out with a low codec rate. After an initial measurement period and RTCP receiver reports indicating a “good channel,” the sending side will attempt to increase the codec rate. The same procedure is executed after a codec rate reduction.

Figure 6.8 depicts how codec rate reduction works in Rel-9:
  • Step 0. The SIP session is negotiated with the full set of codec rates and independent of network level congestion. The use of ECN has to be negotiated separately for each media stream (e.g. VoIP).
  • Steps 1 and 2. After ECN has been successfully negotiated for a media stream the sender must mark each IP packet as ECN-Capable Transport (ECT). Two different values, 10 and 01, have been defined in an IETF RFC106 to indicate ECT. However, for MTSI only 10 shall be used.
  • Step 3. To free up capacity and allow more VoIP calls and/or to improve VoIP coverage, the eNodeB sets the ECN field to Congestion Experienced (CE) in an IP packet that belongs to an IP flow marked as ECT. Note that the ECN-CE codepoint in an IP packet indicates congestion in the direction in which the IP packets are being sent.
  • Steps 4 and 5. In response to an ECN-CE the receiving MTSI client issues an RTCP message to trigger a codec rate reduction.
Note that ECN operates in both directions (uplink and downlink) entirely independent and without any interactions. It is very well possible to trigger codec rate adaptation in one direction without triggering it in the other direction.


A new work item called, Enabling Encoder Selection and Rate Adaptation for UTRAN and E-UTRAN, has been created for 3GPP Rel-10. Part of this work item is to extend the scope of the codec rate adaptation solution agreed in Rel-9 to also apply to HSPA and non-voice RTP-based media streams.

Further Reading:

Thursday, 3 December 2009


Nokia publicly underlined its commitment to broadcast-mobile-TV standard DVB-H with the recent unveiling of the mobile TV edition of the Nokia 5330 and its pretax, presubsidy price tag of €155 (US$230), after some in the industry had questioned its enthusiasm for launching new DVB-H devices. Nokia also quelled any suggestions that it might start supporting the MBMS standard with its future device launches.

The price is a massive drop from the €550 price tag carried by Nokia’s last fully DVB-H-compatible handset, the N96, which launched in 3Q08. So the official line from Nokia is this: “All is well on the good ship DVB-H.”

Read more here.

Meanwhile, In China, China Unicom has launched 3G telecom services in 268 cities across the country, said Li Gang, another deputy general manger for Unicom Group, noting that the WCDMA network supports a 14Mbps download data transmission speed and a 7.2Mbps upload data transmission speed.

Notably, the carrier has adopted the most advanced R6 technology in its core WCDMA network to smooth a WCDMA-to-EPS migration in the future, according to Mr. Zhang.

The China Unicom network is expected to support MBMS and HSPA+64QAM technology in the first phase of a further evolution, shore up a HSPA+MIMO technology in the Phase II evolution, and prompt a LTE technology in the Phase III evolution, said Mr. Zhang, adding that the network will present a 100Mbps download speed and a 50Mbps upload speed after the Phase III evolution.

Read more here.
Back in September, Orange Moldova announced the launch of the world's first mobile telephone service offering high-definition (HD) sound. The service will provide customers with a significantly improved quality of service when making calls. Unlike for other mobile technologies such as multimedia capabilities, this is the first time since the 1990s that mobile voice technologies have been subject to a significant evolution.

This is the second step in Orange’s HD voice strategy, following on from the launch of a high-definition voice service for VoIP calls in 2006. Over 500,000 Livephone devices have already been sold in France and the range will be extended to other Orange countries over the coming months.

The first mobile handset integrating high-definition voice capability that will be launched by Orange Moldova is the Nokia 6720c. This innovative handset integrates the new WB-AMR technology, which is widely expected within the industry to become a new standard for mobile voice communications.

Thanks to the Adaptive Multi Rate-WideBand (AMR-WB) codec, double the frequency spectrum will be given over to voice telephony over traditional voice calling. Orange boasts that the result is "near hi-fi quality" and "FM-radio quality", which seems an odd comparison.