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CCU-N-GML is a new version of the X86 motherboard for communication systems, using Intel quad-core processor, CCU-N-GML can simultaneously handle up to 450 concurrent calls of G.711 codec or 200 concurrent calls of G.729 codec. To bring users clear, high-fidelity audio and video calls, the CCU-N-GML integrates rich HD voice and video codecs. The CCU-N-GML supports comprehensive protocol processing, supporting SIP, IAX2 and other protocols on the IP side and BRI, PRI, SS7, R2, GSM, WCDMA and other protocols on the CPE side. The CCU-N-GML can support 3CX, Asterisk, Issabel, FreePBX, VitalPBX and IPPBX/IVR and other open source software applications as well as PBX, firewall, IVR and voice gateway applications.
•Transport Protocol: UDP, TCP, TLS, SRTP
•DTMF: In-band, RFC2833, SIP INFO
•IP Services: Static IP, DHCP, VPN, Firewall, PPPoE, Bridge
•Support SIP and IAX2
•Web-based operator panel
•DISA (Direct Inward System Access)
•Call detail record (CDR) report
•Call recording, Call parking, call queues, Voicemail, Conference
•Echo canceller
•Callback support
•Flexible and configuration IVR
•Support for PIN sets
•Support for time conditions
•Support extension following
•Support ringing groups
•Support for video calls
•Channel usage report
Figure 1 CCU-N-GML Panel Appearance
1.Testing Platform
CPU |
Main Frequency |
Memory |
Storage |
Operating System |
Asterisk Version |
BIOS |
Intel(R) Celeron(R) N4120 Quad-Core Quad-Thread |
1.10GHz |
8G |
MSATA 16G |
debian9.8 Kernel:4.9.198 |
17.8.0 |
2021.07.21 |
Related notes:
1. Equipment with: Pentium N4120 motherboard through the sip relay docking test machine
2. Test data: test different codes, the maximum number of concurrent sip calls in the case of normal voice
3. Call rate: 1min per call round, 10s internal call delay, maximum number of initiated and stopped calls per second set to 30
4. Asteirsk using source code compilation, g723、g729 coding module using core2-sse4 64-bit so file
5. The performance screenshot is the output of WinSIP call hitting full moment using top command.
6. Modules.conf file was modified during testing, chan_sip was used, pjsip was not used.
Calling process:
sip relay
Winsip --> Server A ( N4120) ------------->Server B(playback demo-instruct)
2.Test method
Prepare a test server B with Asterisk, WinSIP initiates a call to A (N4120) motherboard, which is relayed to Server B for playback via SIP trunk, and make the CPU in full speed working state before each test.
Set/sys/module/pcie_aspm/parameters/policy to performance, and use cpupower -c all frequency-set -g performance to make the CPU in full speed working mode.
CPU operating frequency:
cpupower -call frequency-info |grep "current CPU frequency"
3.Test results
Coding |
Number of concurrent |
Sound Quality |
Performance(Cpu Idle) |
Memory Usage(Mb) |
ulaw |
450 |
Clear voice with no noticeable lag |
0.3% |
597 |
g723 |
250 |
Clear voice with no noticeable lag |
1.7% |
565 |
g729 |
240 |
Clear voice with no noticeable lag |
1.6% |
497 |
4.Notes
1. In the test, the number of concurrent calls is gradually increased until the voice stutters after the call is full to determine the critical number of concurrent calls.
2. In the test to monitor the human ear to judge the sound quality, there are subjective differences, call the full softphone and then call all the way to listen to the voice, the voice is clear and no lagging phenomenon as the standard;.
3. Test process BIOS inside the power limit option has been removed, the test cpu in full speed work state.
Parent topic: Core Control Unit