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Test cases proposal and description of NGN Testbed

Xueli Zhangzhangxueli@mail.ritt.com.cn

CJK 7th NGN-WG MeetingOct10-11, Tokyo, Japan

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CCSA CJK Testbed Team List

　

No.

Name Role Telephone Email

1 Zhang　Xueli project　manager　and　coordinator 86　10　68094404 zhangxueli@mail.ritt.com.cn

2 Zhang　Dehua project　manager　and　coordinator 86　10　62302432 zhangdehua@mail.ritt.com.cn

3 Liu　Shu Testbed　architecturing　and　interconnection86　10　62302900

liushu@mail.ritt.com.cn

4 Zhou　Yi Test　Specification　and　test　86　10　68094138

zhouyi@mail.ritt.com.cn

5 Lin　Meiyu Test　Specification　and　test　86　10　68094152

linmeiyu@mail.ritt.com.cn

6 Xue　Ning Test　Specification　and　test　86　10　62303460

xuening@mail.ritt.com.cn

7 Han　Baoyu Lab　environment　and　test86　10　62302435

hanbaoyu@mail.ritt.com.cn

8 Li　Cheng test　86　10　68094265

licheng@mail.ritt.com.cn

9 Qu　Zhenhua test 86　10　68094265 quzhenhua@mail.ritt.com.cn

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RTNet’s CS Architecture

IP Network

Lanswitch

FEPSTN

SoftX3000

SIGTRAN

U-NICA

MRS6100

IOSS

OpenEye

SoftPhoneVedioPhone

IADIAD

STUN

Lanswitch

SCP

SG7000

STP

UMG 8900+UA5000

POTS　/　BRI/ADSLPOTS　/　BRI/ADSL

…

Lanswitch

Eudemon

MGCP

UMSN2000

SIP/MGCP/H.248/H.323 SIP

MGCP

H.248

ISUP

INAP

FTP/FTAM

E1

SNMP

USAU

ISUP

SIGTRAN

SIP

SIPH.323

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RTNet’s CS Architecture

• 1. SoftX3000, the Softswitch equipment, which fulfill call control, access control of media gateway, distribute resources, process the protocol, routing and accounting, etc. SoftX3000 provide API for the third part applications, and use SIP or SIP-I protocol with other softswitch. The IP address of SoftX3000 is 172.24.253.30. The phone numbers of those in SoftX3000 are 9200XXXX.

• 2. iOSS , integrated Operation Support System, which includes integrated network management system for NGN equipment managing and accounting system.

• 3 ． SCP, Service Control Point, which is the core of the intelligent network, and store user data and service logic, control SSP for service action.

• 4 ． MRS6100, Media Resource Server, process media for services, such as, provide service tone, IVR, some advanced audio service, etc.

• 5 ． U-NICA, the Application Server, which responsible for the logic creation and management of those value-added service and intelligent network service, the Application Server can provide API for a third part development.

• 6 ． SG7000, Signaling Gateway, which fulfills transformation between No.7 signaling in PSTN and signaling in IP network (such as SIGTRAN).

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RTNet’s CS Architecture • 7 ． USAU, Universal Signaling Access Unit, which makes signali

ng transformation, especially transformation between INAP and corresponding signaling in IP network.

• 8 ． UMS N2000, the integrated network management system. • 9 ． Eudemon2100, the session border controller, help signaling an

d media message traverse NAT and firewall. • 10 ． UMG8900, Universal Media Gateway, which is mainly a trun

king gateway, with SG and AG function inside. The encoding and decoding arithmetics UMG8900 supported are G.711, G.723.1, G.726, G.729.

• 11 ． UA5000, an integrated access device, which provides interfaces to different kinds of users, such as analog users, ISDN users, xDSL users, etc.

• 12 ． IAD, Integrated Access Device, access data, audio and video services to IP network for user, and support G.711 arithmetic.

• 13 ． openEye, which is a SoftPhone using SIP protocol, and support G.711, H.261 and H.263 arithmetic.

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Test Cases Proposal

• SIP and PSTN user are the common part for CCSA RTnet and ETRI– Basic voice call

• SIP phone – SIP phone – Normal call– Abnormal call-the called party is busy– Abnormal call- no answer– Abnormal call- the number doesn’t exist

• PSTN user – SIP phone – Normal call– Abnormal call-the called party is busy– Abnormal call- no answer– Abnormal call- the number doesn’t exist

– Multimedia call• SIP phone – SIP phone

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Test description TEST NUMBER:：

TITLE：Basic Call between SIP users

SUBTITLE： normal call

PRE-TEST CONDITIONS：

1、SIP user 1 registered on SS1; SIP user 2 registered on SS2.

TEST CONFIGURATION：SIP1－SS1－SS2－SIP2

TEST STEP：

1、User1 calls user2，user2’s phone rings；

2、User2 answers the call；

3、User1 hangs up first,then user2 hangs up,too.

EXPECTED SIGNAL SEQUENCE:：

SS1 SS2 SI P 2

I NVI TE

100 Tryi ng

(1)

SI P 1

I NVI TE (2)(3)

I NVI TE (4)100 Tryi ng (5)

180 Ri ngi ng (6)

200 OK (7)180 Ri ngi ng

200 OK (8)180 Ri ngi ng

200 OK(9)

ACK(10)ACK

ACK

Medi a Sessi on

(1)SIP user1 sends an INVITE request to SS1 to setup a session，SS1 responses

100 Trying.SS1 forwards the INVITE request to SS2 then to user2.

(6) SIP user1 rings .User 2 sends a 180Ringing response to SS2,SS2 forwards

this message to SS1 then to user1.

(9)User2 answers the call ,User 2 sends a 200 OK response to SS2 then to User1.

(10)User1 sends ACK back.

User1 and user2 setups a multimedia session.

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Test description

• Voice Codec• ☐ G.711, G.729A, G.723.1• ☐ Codec Characteristics•  o. G.711(u/a)•    - payload type : 0/8•    - clock rate : 8000•    - ptime : 20•  o. G.729A•    - payload type : 18•    - clock rate : 8000•    - ptime : 20•  o. G.723.1•    - payload type : 4•    - clock rate : 8000•    - ptime : 30

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Test description

• Video Codec• ☐ H.263/CIF, H.263/QCIF• ☐ Codec Characteristics•  o. H.263/QCIF•    - payload type : 34•    - frame size : 176*144•    - clock rate : 90000 또는 60000•    - frame rate : 3~30•  o. H.263/CIF•    - payload type : 34•    - frame size : 352*288•    - clock rate : 90000•    - frame rate : 3~30• ☐ Video data is required to be transmitted according to the negotiated

value of the quality (bandwidth, frame rate, etc.)

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Test description • Video Related SDP Specification • ☐ The following video related attributes are mandatory.•  o. Media Type•    - Format: m=video <port> RTP/AVP <payload list>•    - Reference : RFC2327•  o. RTP map•    - Format: a=rtpmap:<payload type> <encoding name>/<clock rate>•    - Reference : RFC2327•  o. Bandwidth •    - Format: b=<modifier>:<bandwidth-value>•    - Reference : RFC2327•  o. Framerate •    - Format : a=framerate:<frame rate>•    - Reference : RFC2327•  o. Framesize •    - Format: a=fmtp:<payload type number> <framesize>=[<xMax>, <yMax>,]<MPI>•    - Reference : draft-ietf-avt-rfc2429-bis-09• ☐ bandwidth is represented per medium• ☐ Framerate in RFC2327 precedes that of draft-RFC2329-bis-09• Route/Record-Route Header• ☐ Record-Route header is Optional