RTP implemented in Abacus

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Moris Derrick Osborne
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1 Spirent Abacus RTP implemented in Abacus 编号版本修改时间说明 1

1. Codec that Abacus supports. G.711u law G.711A law G.726 G.726 ITU G.723.1 G.729 AB (when VAD is YES, it is G.729AB, when No, it is G.

2 1. Codec that Abacus supports. G.711u law G.711A law G.726 G.726 ITU G G.729 AB (when VAD is YES, it is G.729AB, when No, it is G.729A) G.729 B (when VAD is YES, it is G.729B, when No, it is G.729) AMR AMR-WB EVRC EVRC-B ilbc GSM-EFR GSM-FR G.722 G Custom Audio Codec 2

specified, with bit rates: 6.6, 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05, 23.85 Kbit/s 3.

3 (1) AMR- Wide Band 1. Definition: 1. AMR-WB stands for Adaptive Multi-Rates Wideband 2. Based on ITU G722.2, which defines the coding scheme of 16-bit uniform PCM signals Coding scheme used: ACELP (Algebraic Code Excited Linear Prediction Coder) VAD is also integrated into the ACELP 9 codecs are specified, with bit rates: 6.6, 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05, Kbit/s 3. The sampling rate of the speech signal to be processed is 16K samples per second A normal PCM samples the audio signal at 8 K samples per second 16 K samples per second will be able to support a audio signal with bandwidth up to 8 KHz (ITU spec. 7KHz) 4. Applications: VoIP, GSM, 3GPP, ISDN wide band telephony 2. Abacus s implementation: 1. This codec can not be combined with other codec in the RTP protocol development, supported VoIP protocols: SIP, MGCP, Megaco, BICC (H323 not supported.). 2. Only allows original wave files sampled at 16 K/s,System will flag compilation error if wave files sampled at 8K/s is selected. 3

4 3. Requires two new option: AMR-WB Encoding and AMR-WB Decoding 4. Support Decoding on certain path confirmation types: Types supported: PESQ and Voice File Matching Types NOT supported: PSQM, Echo, Tone and T30Fax 5. ARM-WB in SDP Media description: 4

(2) EVRC-B 1. EVRC: Enhanced Variable Rate Codec: Part of 3GPP Project, Rates can be adjusted dynamically during the call, Allow bandwidth adjustment by Service Provider based on traffic situation. 2.

5 (2) EVRC-B 1. EVRC: Enhanced Variable Rate Codec: Part of 3GPP Project, Rates can be adjusted dynamically during the call, Allow bandwidth adjustment by Service Provider based on traffic situation. 2. EVRC-B: Enhancement to EVRC: Use a different encoding algorithm (Noise Excitation Linear Prediction NELP); Better QOM than EVEC when using the same rate ; Compresses 20ms samples of 8000 Hz, 16 bits speech into 4 different rates: Rate1: 171 bits, Rate 1/2: 80 bits, Rate 1/4: 40 bits (rate only supported in EVRC-B, not EVRC, used mostly for noise/ silence)), Rate 1/8: 16 bits (used mostly for noise silence). 3. Abacus s Implementation: (1) Need to choose QOM with 16bit 8KHz sample wave files (2) Supports both Encoding and Decoding: Expects the QOM rating tends to be lower, due to the high compression. (3) Supports SIP, Megaco and Clear Channel (4) Support both ICG and A50E (5) Require options 4. Codec Groups 1. VoIP Compression Group A (Encoding/Decoding): G711 A-Law, G711 mu-law, G.723.1, G.726, G.726 ITU,G.729AB 2. VoIP Compression Group B (Encoding/Decoding): G711 A-Law, G711 mu-law, G.729B, AMR, EVRC 3. VoIP Compression Group C (Encoding/Decoding): G711 A-Law, G711 mu-law, AMR-WB 4. VoIP Compression Group D (Encoding/Decoding): G711 A-Law, G711 mu-law, EVRC-B Notes: For A50 Eth You can only select codecs within the same group into your RTP protocol. This is because it has only one DSP. No such limitation for ICG. 5

6 (3) G722 Codec (4) G722.1 Codec Note: Abacus Implementation: 1. Requires Option, Configurable in RTP Protocol Development and Supports both Coding and Decoding, Only Supports SIP, MGCP and Megaco 2. H323 not supported, PSQM also not supported. 6

Audio signals sampled at 8 Khz/13 bits 4. Provide toll quality voice coding / decoding Abacus Implementation: 1.

7 (5) GSM-EFR (Enhanced Full Rate) Standardized by ETSI-GSM for wireless and VoIP use (ETSI-06.10) 1. 1Use RTP LTP (Regular Pulse Excitation Long Term Prediction) for coding 2. Bit Rate: 13 Kbps 3. Audio signals sampled at 8 Khz/13 bits 4. Provide toll quality voice coding / decoding Abacus Implementation: 1. Requires Option 2. Configurable in RTP Protocol Development 3. Supports both Coding and Decoding 4. Limitation: H323, Sknny, E-model Not supported (6) Summary: DSP Set up 7

2. 2833 Feature RFC 2833 describes the passing of telephony signal using RTP,

8 Feature RFC 2833 describes the passing of telephony signal using RTP, This includes DTMF, onhook, off-hook, ring and Trunk Signals using ABCD bits. 8

3. SRTP usage. (1) Based on Two Standards: 1. RFC 3711: The Secure Real-Time Transport Protocol : Define the cryptographic schemes used to secure RTP and New RTP and RTCP packet format 2.

materials, Applicable to SDP based Signaling Protocols like SIP, MGCP, and Megaco. (2) New attribute in SDP is created to support the communication and negotiation of the crypto parameters and key.

9 3. SRTP usage. (1) Based on Two Standards: 1. RFC 3711: The Secure Real-Time Transport Protocol : Define the cryptographic schemes used to secure RTP and New RTP and RTCP packet format 2. Draft RFC: Draft-ietf-mmusic-sdescripitons: Session Description Protocol Security Descriptions : This draft RFC recommends the use of SDP for the communication of the cryptographic schemes and key materials, Applicable to SDP based Signaling Protocols like SIP, MGCP, and Megaco. (2) New attribute in SDP is created to support the communication and negotiation of the crypto parameters and key. a = crypto :<tag> <crypto suite> <key parameters> [<session parameters>] 1. tag: ID for the key exchange negotiation 2. crypto-suite: ID representing the encryption and authentication algorithms 3. key-parameters = inline : key information 4. Session-parameters: optional, define the transport 9

10 (3) Abacus Supports SRTP on SIP, MGCP and Megaco. 10

4. Media Loopback (1). Base on a draft rfc: draft-ietf-mmusic-media-loopback-10 1. The objects of test include: Voip, Text and Video 2.

11 4. Media Loopback (1). Base on a draft rfc: draft-ietf-mmusic-media-loopback The objects of test include: Voip, Text and Video 2. Use of Loopback at the user side to reflect back the send data 3. By Measuring the data reflected back, the quality of the media path can be assessed. (2) Background Information 1. An active test to get the performance (compared with the passive monitoring method) 2. The service provider is the source, always the one sending and receiving the data. 3. The Subscriber is the mirror, always the one receiving and re-sending the data 4. The test is part of a special test call that can be originated from the Service Provider or the Subscriber. With that, there can be 2 scenarios: Service Provider (Source) originating --- Subscriber (Mirror) terminating Service Provider (Source) Terminating - Subscriber (Mirror) Originating 5. Different Options of the test Packet only loop back (Basic test on the media path) Media loop back (Closer to the analog interface) 11

12 6. Use of SDP to execute the loopback, New media attributes: a=<loopback-mode> #<loopback-mode> can be: loopback-source #sender of the message is the source of loopback loopback-mirror #sender of the message is the mirror of loopback a=loopback:<loopback-type> #<loopback-type> can be: rtp-pkt-loopback #packet only rtp-media-loopback #media rtp-start-loopback #special loopback attribute to accommodate firewall (3)Abacus s Implementation 1. SIP Only 2. Only voice (Packet-only) is supported. No video. No text. 3. Source Originating is the only required scenario. But in reality, Abacus supports both source originating and source terminating 4. Configuration: In Channels/PC/Advanced/QOM and In RTP/New tag of media 12

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5. RTCP for TuroRTP (1) Background Information 1. Existing TurboRTP only sends out RTP, with NO RTCP being sent at the same time 2. This feature is to add RTCP to the TurboRTP feature 3.

15 5. RTCP for TuroRTP (1) Background Information 1. Existing TurboRTP only sends out RTP, with NO RTCP being sent at the same time 2. This feature is to add RTCP to the TurboRTP feature 3. Requires a new option on ICG (2) Abacus Configuration 1. In Protocol Development /RTP /MISC /Check the Use RTCP with TurboRTP Scripts box. 2. Apply the new RTP protocol in P&T /Protocols /TP Protocol (3) New Variances Created: 1. RTP Packet Loss (per RTCP packet) 2. RTP Jitter (msec, per RTCP packet) 3. RTP packet loss rtate Existing Variances Changed 1. RTP Packet Loss (per check interval) RTP Packet Loss (per check interval, TuroRTP) 2. RTP Jitter (msec) RTP Jitter (msec, TurboRTP) 15

6. RTP Replay (1) Background Information Abacus supports a number of codecs for VoiP testing already,there are from times to times new codecs coming up, existing codecs being enhanced, or proprietary

16 6. RTP Replay (1) Background Information Abacus supports a number of codecs for VoiP testing already,there are from times to times new codecs coming up, existing codecs being enhanced, or proprietary codecs brought to the place.this means Abacus has to add these codecs if we are to support them. This takes time and efforts, and might not be able to meet customers needs.to overcome these odds, this feature is to allow users to create their own RTP packet stream, and use it as source files to facilitate testing. (2) Abacus s Implementation For SIP only Support Packet only path confirmation Valid for Audio and Video RTP streams Customer has to provide his own RTP media streams Note: There are very specific requirements on the stream: Must be one single media stream from the same source and destination (defined by source IP + port and Destination IP + port) Must be in.pcap format Consists of RTP packets only (not even RTCP) All packets must be of the same payload type GUI will check if these requirements are met (3) Abacus configure New Packet Only Path Confirmation for PCAP file New Custom Audio Codec and Custom Video Codec in the Protocol Selection RTP 16

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Note: for users configuring custom codec, the RTP Replay function will NOT support the following functions: Send Video / Audio Wait QOM Video / Audio Path Confirmation The codec and its related

18 Note: for users configuring custom codec, the RTP Replay function will NOT support the following functions: Send Video / Audio Wait QOM Video / Audio Path Confirmation The codec and its related parameters configured in the GUI MUST be the same as that in the RTP media steam 7. Abacus Capacity for v6.00 Note: SIP Turbo RTP at 20mS packet size Increased from 7,168 to 10K channels (single IP, G.711) Increased from 7,168 to 12K channels (single IP, EVRC/EVRC-B) R-Factor (G.107) Increased from 3,000 to 4,096 channels Audio packet path confirmation Increased from 1,024 to 2,048 channels 18

VoIP Basics. 2005, NETSETRA Corporation Ltd. All rights reserved.

VoIP Basics. 2005, NETSETRA Corporation Ltd. All rights reserved. VoIP Basics Phone Network Typical SS7 Network Architecture What is VoIP? (or IP Telephony) Voice over IP (VoIP) is the transmission of digitized telephone calls over a packet switched data network (like