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6 Media Encoding and Transport

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6.1 Overview ........................................................................................ 3 6.2 Adaptive Multi-Rate Codec (1/4).....................................................4

6.2 Adaptive Multi-Rate Codec (2/4).....................................................5 6.2 Adaptive Multi-Rate Codec (3/4).....................................................6 6.2 Adaptive Multi-Rate Codec (4/4).....................................................7 6.3 AMR-WB Codec (1/4)..................................................................... 8 6.3 AMR-WB Codec (2/4)..................................................................... 9 6.3 AMR-WB Codec (3/4)................................................................... 10 6.3 AMR-WB Codec (4/4)................................................................... 11 6.4 Video Codecs (1/2).......................................................................12 6.4 Video Codecs (2/2).......................................................................13

6.5 H.263 Codec (1/3) ........................................................................14

6.5 H.263 Codec (2/3) ........................................................................15 6.5 H.263 Codec (3/3) ........................................................................16 6.6 H.261 Codec ................................................................................17 6.7 MPEG Codecs (1/3) .....................................................................18 6.7 MPEG Codecs (2/3) .....................................................................19 6.7 MPEG Codecs (3/3) .....................................................................20 6.8 Text Encoding (1/3) ......................................................................21 6.8 Text Encoding (2/3) ......................................................................22 6.8 Text Encoding (3/3) ......................................................................23 6.9 Save Media Transport (1/2) ..........................................................24 6.9 Save Media Transport (2/2) ..........................................................25 6.10 Real-Time Transport Protocol (1/4) ............................................26 6.10 Real-Time Transport Protocol (2/4) ............................................27 6.10 Real-Time Transport Protocol (3/4) ............................................28 6.10 Real-Time Transport Protocol (4/4) ............................................29 6.11 Real-Time Transport Control Protocol (1/3)................................30 6.11 Real-Time Transport Control Protocol (2/3)................................31 6.11 Real-Time Transport Control Protocol (3/3)................................32

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6.1 Overview

In this module we will have a closer look how IMS media, i.e., speech, video and text areencoded to assure a high quality when transmitted over the air interface.

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6.2 Adaptive Multi-Rate Codec (1/4)

3GPP defined two speech codecs for IMS terminals: AMR (Adaptive Multi-Rate) speechcodec is the mandatory speech codec. It co-operates with legacy GSM terminals usingEnhanced Full-Rate speech codecs. Then we have the Adaptive Multi-Rate Wide-Bandspeech codec for those UEs supporting wide-band services.

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6.2 Adaptive Multi-Rate Codec (2/4)

AMR consists of eight different codecs, each with a different bandwidth. Their differentbandwidths are 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, and 4.75 kbps. These are alsoreferred to as AMR modes. The 12.2, 7.40, and 6.70 kbps AMR modes are also known asGSM-EFR (Enhanced Full Rate) codecs.

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6.2 Adaptive Multi-Rate Codec (3/4)

The AMR modes were designed initially to be used in GSM networks that provide a fixed ratefor circuit-switched voice calls. This rate is split into channel coding and speech coding.When radio quality is poor due to high air interface interference the terminals use low-bandwidth AMR modes, e.g., 4.75 kbps well protected by a large channel coding element.When the air interface interference is low terminals use high AMR bandwidth modes with12.2 kbps and only a few bits for channel coding.

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6.2 Adaptive Multi-Rate Codec (4/4)

The AMR codec itself is able to switch modes on a frame-by-frame basis. That is, one 20 msspeech frame can be encoded using one particular AMR mode and the next speech frameusing another one.

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6.3 AMR-WB Codec (1/4)

3G networks using WCDMA access do not use AMR modes in the same way as GSMnetworks. WCDMA uses fast power control and does not perform mode adaptation at thechannel-encoding level. WCDMA networks use low-bandwidth modes to gain capacity whenmany users make voice calls at the same time.

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6.3 AMR-WB Codec (2/4)

When AMR is transported over a packet-oriented path the overhead introduced by the RTP,UDP and IP headers is fairly large. Thus, it only allows for low-bit rate speech coding. IPheader compression can be used to gain some extra payload capacity to use high-bandwidth

AMR modes.

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6.3 AMR-WB Codec (3/4)

The AMR-WB codec as defined in 3GPP Technical Specification 26.190 encodes voice using16,000 samples per second, instead of 8,000 samples per second used by AMR. This highersampling frequency allows AMR-WB to encode a wider range of frequencies. Consequently,

AMR-WB encodes speech with higher quality than the codecs we described earlier.

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6.3 AMR-WB Codec (4/4)

AMR-WB consists of a family of codecs which encode audio using the following bandwidths:23.85,23.05, 19.85, 18.25, 15.85,14.25, 12.65, 8.85, and 6.60 kbit/s. AMR-WB is themandatory codec for 3GPP IMS terminals that provide wideband services. Thus, IMSnetworks provide better speech quality due to higher bit rates per speech sample for bothhigh and low air interface interference situations.

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6.4 Video Codecs (1/2)

Video encoding relies on single image encoding. Video transmission uses a set of encodedstill images taken with a short interval between them. If the subsequent presentation of thestill image is fast enough the human eye perceives this succession of images as a movingimage. A standard sequence used in TV systems issues 25 pictures per sec, i.e., each 40msa new picture is presented.

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6.4 Video Codecs (2/2)

IMS video encoding formats are well defined by international standardization bodies.Examples include: ITUs H.263 which is mandatory for IMS or H.261 an additional option.MPEG (Motion Picture Experts Group) developed several standards that can also be usedfor IMS services.

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6.5 H.263 Codec (1/3)

H.263 evolved out of H.261, the previous ITU standard for video compression and theMPEG-1 and MPEG-2 standards.

H.263 was developed for the ITU-T H.324 multimedia framework used with low bit ratecommunications. It supports five formats to encode images: Sub-QCIF (Quarter CommonIntermediate Format), QCIF, CIF, 4CIF and 16CIF. All of these formats encode the color ofthe pixels using a luminance component and two chrominance components, but supportdifferent resolutions.

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6.5 H.263 Codec (2/3)

The luminance components is the black-and-white component of the image and defines howdark or light each pixel is. The chrominance components provide the color of a pixel inrelation to the colors red and blue.

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6.5 H.263 Codec (3/3)

Image resolution formats used by H.263 are related to the four defined formats and vary inresolution. Luminance resolution can vary between 128 x 96 pixels and 1408 x 1152 pixels.The chrominance resolution varies between 64 x 48 pixels and 702 x 576 pixels.Please note the 2:1 relationship between the figures. This results from the fact that thehuman eye is more sensitive to luminance information than to chrominance information.

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6.6 H.261 Codec

ITU-Ts H.261 standard defines the video codec used by the H.320 video-teleconferencingframework. This codec was designed to be used over ISDN lines and therefore, producesbandwidths that are multiples of 64kbps ranging from 64 to 1984 kbps.

The data rate of the coding algorithm was designed to be able to operate between 40 kbpsand 2 Mbps. The standard supports CIF and QCIF video frames with luminance resolutionsof 352x288 and 176x144 respectively. The chrominance resolutions are a 176x144 and88x72. It also has a backward-compatible trick for sending still picture graphics with 704x576luminance resolution. This was added in a later revision in 1994.

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6.7 MPEG Codecs (1/3)

The MPEG video standards are used for both media storage and for videoconferencing.MPEG-1 was developed to encode audio and video at rates about 15Mbps. It also includesthe popular Layer 3 (MP3) audio compression format.

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6.7 MPEG Codecs (2/3)

MPEG-2 was developed to encode audio and video at rates between 4 and 80 Mbps which ishigher than MPEG-1, also providing higher quality. While the VCD (Video CD) format isbased on MPEG-1, DVD (Digital Video Disk), DVB (Digital Video Broadcasting) and somedigital satellite and cable television systems are all based on MPEG-2.

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6.7 MPEG Codecs (3/3)

Initially, MPEG-3 was designed for High-definition TV, but it was abandoned when it wasdiscovered that MPEG-2 with extensions was sufficient for HDTV. MPEG-4 expands MPEG-1 to support video / audio "objects", 3D content, low bitrate encoding and Digital RightsManagement. The DivX and XviD formats are based on MPEG-4.

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6.8 Text Encoding (1/3)

Text already consists of digital information, so there is no need to perform the analog-to-digital or digital-to-analog conversions which are needed for audio and video. There are twotypes of textcommunication: Instant messages and real-time text.

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6.8 Text Encoding (2/3)

Instant messages convey a whole message, such as: "How are you?" That is, the sendertypes the message, edits it if necessary, and sends it. This way the receiver only gets thefinal version of the message.

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6.8 Text Encoding (3/3)

Real-time text consists of transferring keystrokes instead of text. If the sender writessomething and then deletes it to write something else, the receiver will see how thesechanges are performed. That is, the receiver gets letters and commands (e.g., carriagereturns or delete characters) one by one as they are typed. The most common format forreal-time text is ITU-Ts Recommendation T.140.

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6.9 Save Media Transport (1/2)

For a safe media transport, the critical question is whether or not the payload data cantolerate a certain amount of packet loss or not. Audio and video messages will loose qualitydramatically if packets are lost whereas web browsing or instant messaging are more robustapplications.

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6.9 Save Media Transport (2/2)

Hence, the type of media should choose within IMS depending on the question whether it isgoing to be transported on reliable connections like TCP or unreliable ones like UDP.

As UDP is widely used to transport all kinds of media in IP networks some additionalmeasures must be taken to ensure a proper media transport. The IETF has specified theReal-time Transport Protocol and its sister protocol, the Real-time Transport Control Protocolto reside upon UDP for this purpose.

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6.10 Real-Time Transport Protocol (1/4)

Applications using RTP are less sensitive to packet loss, but typically very sensitive todelays, so UDP is a better choice than TCP for such applications. Services provided by RTPinclude:

Payload-type identification - Indication of what kind of content is being carried Sequence numbering - PDU sequence number Time stamping - presentation time of the content being carried in the PDU Delivery monitoring - packet can still delivered out of order

RTP does not provide mechanisms to ensure timely delivery.

Nor does it give any Quality of Service (QoS) guarantees. These things must be provided by some other mechanism.

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6.10 Real-Time Transport Protocol (2/4)

RTPs major benefit is to allow receivers to play out media at a proper pace even if the IPnetwork does not keep the time relationship of the transported data.

As we know IP networks can produce jitter, that allows the jitter signal to arrive earlier thanthe original signal. Thus, another scheme is needed by the receiver to re-establish a certaininformation order - this is provided by the RTP timestamps.

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6.10 Real-Time Transport Protocol (3/4)

The receiver stores all incoming data in a buffer according to their timestamps and thenstarts playing them. If a certain packet is still missing because of delay an interpolation of thepresent signal is played instead. It might even be a simple replay of the whole sample. If thepacket arrives afterwards it is discarded.

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6.10 Real-Time Transport Protocol (4/4)

t is obvious that the receiver should not wait too long before starting to play, nor should itstart too early. Field trials have proven that a majority of bits arrive 50ms after they weresent. A waiting time of approx. 100ms should be a good interval before packets are played.

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6.11 Real-Time Transport Control Protocol (1/3)

RTCP, which stands for Real-time Transport Control Protocol, provides out-of-band controlinformation for RTP flow. It partners RTP in the delivery and packaging of multimedia data,but does not transport any data itself.

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6.11 Real-Time Transport Control Protocol (2/3)

It is used periodically to transmit control packets to participants in a streaming multimediasession. The primary function of RTCP is to provide feedback on the quality of service beingprovided by RTP.

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6.11 Real-Time Transport Control Protocol (3/3)

RTCP gathers statistics on a media connection and information such as the bytes sent, thepackets sent, lost packets, jitter, and round trip delay.

An application can use this information to increase the quality of service perhaps by limitingflow, or using a low compression codec instead of a high compression one.