Research on VoIP test model

IP Voice (voice over Ip,voip) applications more and more popular, common VoIP test model applications include: Test VoIP gateway, VoIP PBX, Gateway Controller (Gatekeeper), proxy server, media Gateway Controller, Soft switches and other internetwork gateways and WAN devices, VoIP conferencing telephony tests. Identify development capacity, functionality, performance, interoperability, and features. Test the interface between a traditional telecom network and a new packet-based network. proves the function and capacity of call accounting, voice messages, and conferencing servers. The VoIP test model provides an Ethernet port for generating and terminating VoIP signaling and transport streams.

At present, the telecommunication network based on circuit switching is evolving to the IP network based on packet switching. SoftSwitch technology, Next Generation Network (NGN) and IP Multimedia Service subsystem (IP multimedia subsystem,ims), which support fixed access and mobile access in the framework of NGN, are all based on IP as core business, meanwhile, for a long time, Voice will still be the core business of telecommunications, while the study of Voice test parameters based on IP network and the implementation of the model is the active exploration under this background [1].

1. VoIP Model characteristic parameters

1.1 VoIP test model should have the common characteristics [2-8]

(1) Related physical properties. There should be multiple standard Ethernet ports, each with link/active and speed LEDs displayed.

(2) VoIP main test index characteristics. Using PESQ to perform the voice quality test for each call; can send and receive signal sound, voice, detection and forwarding DTMF signal sound, can be specified in the test to simulate the specific voice IP network damage, including delay, jitter, chaos, packet loss and other tests; Generate and end multiple VoIP calls at the same time on each test model Build a call to the media Gateway Controller, Gateway, or LF switch. Verify that the voice path is established and maintained during the duration of the call. Measurement delay, call establishment time, lost packet, disorderly package, jitter, BHCA and call completion, provide test message time tag.

(3) VoIP protocol test characteristics. You should have a Session Initiation Protocol (Sessions Initiation PROTOCOL,SIP), Media Gateway Control Protocol (medium Gate controls PROTOCOL,MGCP), H.323, or Media Gateway control (Media Gateway CONTROL,MGACO) protocol generates VOIP call test function and provides two or three-layer line-speed flow test for IP network.

(4) Two times test development platform features. Support Multi-channel VoIP flexible call sequencing test, based on product testing needs, flexible customization protocol, support TCL script automatic testing, automatic and continuous acquisition of test results.

(5) Perfect test report results. The model should produce detailed call error reporting, including an unsuccessful order or information. and use tables and graphs to display the results, listen for arbitrary channels for validation or analysis, and filter the displayed protocol information with the information type or information header.

1.2 VoIP test Performance parameter significance [9]

1.2.1 Delay

Delay refers to the data from the source to the end of the time required, for interactive voice communication systems, the increase in time delay will allow the speaker to feel the speaker hesitate, will also cause echoes, so for VoIP systems, the general control of the delay in 150ms. In VoIP system, the time delay is generally composed as follows: (1) packet encapsulation and sending delay. Packet packaging delay is mainly composed of 2 parts, the 1th part is the speech codec of A/D and d/a transformation, mainly by the equipment hardware, the delay is small, mainly related to the following section of the input signal preprocessing, automatic correction analysis technology; The 2nd part is determined by the number of voice bytes encapsulated by the RTP packet. Packet sending delay refers to the time required for the serial transmission of the packet data to the physical link, which is determined by the packet length and the link transfer rate. (2) Propagation delay. Propagation delay refers to the delay caused by the length of the physical carrier by the physical carrier (copper wire or optical fiber) of the VoIP signal. (3) Queuing and forwarding delay. Queueing and forwarding delay is the important component of the total transmission delay, which is the queue of packet data in the IP network node (switch or router, etc.), and the delay of forwarding from one port to the other end.

1.2.2 Jitter

Unlike the traditional PSTN network, PSTN is a fixed rate (64KBIT/S) for data transmission, and IP packets between the choice of routing, and different routes have different delays and other factors, resulting in the same VoIP packets between different delays, resulting in jitter. In VoIP, as the time delay is not completely eliminated, can only be controlled within a certain range, generally through the device in the buffer to solve, the VoIP performance indicators are required.

2, IP voice flow performance parameters Analysis

IP Voice performance parameters mainly for VoIP voice codec A/d, d/A (such as: g.711, g.723, etc.) [2-8] for analysis, that is, it is for the voice flow analysis, IP voice performance parameters will directly lead to VoIP delay, jitter, voice distortion and so on.

① input signal preprocessing. Input signal preprocessing mainly refers to the IP speech signal before a/D conversion, so that the input signal through the high pass IIR filter processing, can first filter out the undefined frequency of the noise signal, the High-pass filter is defined as

(1)

② Automatic correction analysis. Set the normal automatic correction coefficient for RM, based on the preprocessing signal and the frame energy of 8kHz sampling rate, the corresponding speech A/D conversion automatic correction window is defined as

(2)

③ Reflection coefficient compensation. The reflection coefficient is mainly used in VoIP VAD (mute detection) technology, improve VoIP voice quality. The channel noise can be eliminated by setting a valid threshold. The automatic correction coefficient of mean square error between ordinary and continuous type is compensated according to (3) type.

(3)

(3), if D is less than the adjusted threshold th, and the last frame is inactive, the average coefficient is used for reflection coefficient compensation, otherwise the continuous coefficient rm (i) is used, and the threshold th is determined by the following algorithm according to each frame

This algorithm means the preset mute detection (VAD), then the noise threshold is o, otherwise the noise threshold is calculated according to the coefficient (which is related to the sampling rate and the packet length), and the o.06 is calculated according to the o.06.

④LP Synthesis Linear predictive Synthesis analysis filter [10]. Calculated according to the following excitation:

The reflection coefficient can be converted to the predictive coefficient

　　　　

(4)

(5)

The linear predictive Synthetic analysis filter is defined as:

(6)

The excitation is generated by the filter's comfort noise. L is the length of the excitation, which is usually equal to the frame length. For an inactive frame that follows the 1th active frame, L equals the last frame length minus modulus m. Therefore, the 1th sampled m of the synthesized filter output is often discarded.