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Test cases proposal and description of
NGN Testbed
Xueli Zhang
[email protected]
CJK 7th NGN-WG Meeting
Oct10-11, 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
[email protected]
2
Zhang Dehua
project manager and coordinator
86 10 62302432
[email protected]
3
Liu Shu
Testbed architecturing and interconnection
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Zhou Yi
Test Specification and test
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Lin Meiyu
Test Specification and test
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Xue Ning
Test Specification and test
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Han Baoyu
Lab environment and test
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Li Cheng
test
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Qu Zhenhua
test
2
86 10 62302900
86 10 68094138
86 10 68094152
86 10 62303460
86 10 62302435
86 10 68094265
86 10 68094265
[email protected]
[email protected]
[email protected]
[email protected]
[email protected]
[email protected]
[email protected]
RTNet’s CS Architecture
SCP
SoftX3000
IOSS
FTP/FTAM
SIP
U-NICA
SIP/MGCP
/H.248/H.323
SIP
Lanswitch
SIGTRAN
INAP
USAU
MGCP
MRS6100
STP
IP Network
ISUP
SG7000
ISUP
SNMP
STUN
UMSN2000
FE
PSTN
Eudemon
E1
Lanswitch
Lanswitch
H.248
H.323
SIP
MGCP
UMG 8900+UA5000
POTS / BRI/ADSL
…
VedioPhone
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OpenEye
SoftPhone
IAD
RTNet’s CS Architecture
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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
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7．USAU, Universal Signaling Access Unit, which makes signaling
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 and
media message traverse NAT and firewall.
10．UMG8900, Universal Media Gateway, which is mainly a
trunking 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
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Normal call
Abnormal call-the called party is busy
Abnormal call- no answer
Abnormal call- the number doesn’t exist
• PSTN user – SIP phone
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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:：
SIP 1
SS1
(1)
INVITE
(3)
100 Trying
SS2
INVITE
(2)
100 Trying (5)
180 Ringing
180 Ringing
(9)
200 OK
(10)
ACK
200 OK
SIP 2
(8)
INVITE
(4)
180 Ringing
(6)
200 OK
(7)
ACK
ACK
Media Session
(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.
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User1 and user2 setups a multimedia session.
Test description
• Voice Codec
• ☐ G.711, G.729A, G.723.1
• ☐ Codec Characteristics
• o. G.711(u/a)
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- payload type : 0/8
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- clock rate : 8000
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- ptime : 20
• o. G.729A
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- payload type : 18
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- clock rate : 8000
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- ptime : 20
• o. G.723.1
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- payload type : 4
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- clock rate : 8000
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- ptime : 30
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Test description
• Video Codec
• ☐ H.263/CIF, H.263/QCIF
• ☐ Codec Characteristics
• o. H.263/QCIF
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- payload type : 34
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- frame size : 176*144
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- clock rate : 90000 또는 60000
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- frame rate : 3~30
• o. H.263/CIF
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- payload type : 34
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- frame size : 352*288
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- clock rate : 90000
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- 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
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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
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