Ozeki VOIP SIP SDK

Codec performance test results

The following codec performance test was performed on Ozeki VoIP SIP SDK v9.7.0 on 2012.04.13. The test was setup to verify how many simultaneous SIP/VoIP calls can be established by a standard workstation running an application based on Ozeki VoIP SIP SDK.



The test was executed using the following workstation(s):

Model: Dell Vostro 430
OS: Windows 7 enterprise
Processor: Intel(R) Core (TM) i7 CPU 860 @ 2.80 Ghz 2.80 Ghz
Installed memory (RAM): 4.00 GB
System type: 64 bit Operating System

The software environment consisted of the Ozeki Phone System PBX, and custom test applications based on the Ozeki VoIP SIP SDK Softphone demo. During all tests minimal load was experienced on the PBX system. Hint: The Ozeki Phone System PBX is also based on the Ozeki VoIP SIP SDK. It can be downloaded form www.ozekiphone.com.

During each test "Test Client A" made several calls to "Test Client B", and one call was randomly routed to a "human test point". At each test client an automated voice quality verification module was used to detect system breakdown, by evaluating packet latency and the bit rate after decoding. A call was treated acceptable, as long as sufficient bitrate was produced by the codec to provide continuous audio playback.

voip codec performance test setup
Figure 1 - VoIP codec performance test setup



The test application was used to create as many calls in a loop as fast as possible.

Test results

Codec name Max simultaneous calls limited by CPU Max number of calls on 100MBps LAN Max number of calls on 1000MBps LAN Bandwidth (kbs) packets/s sample interval (ms) Raw data size Enc data size Compression (%)
Alaw 31250 656 6564 78 50 20 320 160 50
Ulaw 36363 656 6564 78 50 20 320 160 50
G722 679 656 679 (limited by CPU)
6564 (theoritical)
78 50 20 320 160 50
G723-30 1374 1374 (limited by CPU)
3103 (theoritical)
1374 (limited by CPU)
31030 (theoritical)
16,5 33,3 30 480 24 95
G726-16000 1811 1641 1811 (limited by CPU)
16410 (theoritical)
31,2 50 20 320 40 87,5
G726-24000 1984 1313 1984 (limited by CPU)
13128 (theoritical)
39 50 20 320 60 81,25
G726-32000 1964 1094 1964 (limited by CPU)
10940 (theoritical)
46,8 50 20 320 80 75
G726-40000 1821 938 1821 (limited by CPU)
9377 (theoritical)
54,6 50 20 320 100 68,75
G728 345 345 (limited by CPU)
1641 (theoritical)
345 (limited by CPU)
16410 (theoritical)
31,2 50 20 320 40 87,5
G729 510 510 (limited by CPU)
2188 (theoritical)
510 (limited by CPU)
21880 (theoritical)
23,4 50 20 320 20 93,75
Speex-narrowband 83 83 (limited by CPU)
1932 (theoritical)
83 (limited by CPU)
19321 (theoritical)
26,5 50 20 320 28 91,25
ILbc-30 173 173 (limited by CPU)
2207 (theoritical)
173 (limited by CPU)
22069 (theoritical)
23,2 33,3 30 480 50 89,58333333
GSM 1059 1059 (limited by CPU)
1829 (theoritical)
1059 (limited by CPU)
18286 (theoritical)
28 50 20 320 33 89,6875

Explanation

Max simultaneous calls: Measured raw codec performance. Theoretical maximum number of calls on this hardware. Measured number of encoding/decoding cycles per second divided by the number of packets to be encoded per second for the given codec. N = 1 second / ((Time required for encoding + time required for decoding) * (number of packets needed for the given codec/second))

Max number of calls on 100MBps LAN: The number of calls that can operate simultaneously if "Test Client A" and "Test Client B" reside on different workstations and they are connected with a 100 Mbps LAN and we use two way audio.

Max number of calls on 1000MBps LAN: The number of calls that can operate simultaneously if "Test Client A" and "Test Client B" reside on different workstations and they are connected with a 1 Gbps LAN and we use two way audio..

Bandwidth (kbs): Bandwidth requirement of a single call (two way audio)

Packet/s: The number of RTP packets transmitted per second to deliver audio.

Sample interval(ms): The length of audio date in a packet

Raw data size: The size of the digitized audio packet before encoding (in octets), assuming 8 khz, 16 bit, mono audio stream.

Encoded data size: The size of the audio packet after encoding.

Compression (%): The compression rate of the codec. The size of the audio packet after encoding divided by the size of the audio packet before encoding

Findings

Each codec requires different CPU processing power for encoding and decoding. For most codecs the maximum number of calls is limited by the CPU processing capability. For high performance systems operating in a LAN environment, we highly recommend the G711 uLaw codec.

During our tests the Ozeki PBX server never experienced high load.