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Product Reviews
February 2003


GIPS VoiceEngine

 

Global IP Sound

900 Kearny St., 5th Fl.

San Francisco, CA 94133

Phone: 415-397-2555

Web: www.globalipsound.com

 

Price: Call for pricing

 

Editor's Choice Award

RATINGS (0-5)
Installation: 5
Documentation: N/A
Features: 5
GUI: N/A (Pocket Presence's GUI was used)
Overall: A-


There are now several VoIP applications designed to run on Pocket PCs that not only enable remote workers to stay in touch, but are poised to threaten cell carriers as well. Global IP Sound (www.globalipsound.com) designs the GIPS VoiceEngine codec, which can be integrated into VoIP applications. Since the codec itself needs to be integrated into an application, they shipped us a third-party application called Pocket Presence, a Swedish-based company that uses their codec, pre-installed on a Compaq iPAQ. VoiceEngine is a software package that handles all the voice components and includes an adaptive jitter buffer (GIPS NetEQ), acoustic echo control, packet loss concealer, and any standard codec can be plugged in, including G.711, Enhanced G.711, and iPCM.

Global IP Sound claims that the GIPS VoiceEngine provides superior voice quality with improved audio processing on the PDA. It does this by minimizing delay, both network (jitter which is very high for WLAN) and delay due to limitations of the PDA (clock drift, real time resource handling, and general OS issues).

In the PocketPC implementation of the GIPS VoiceEngine, for most of our tests we utilized the GIPS iPCM wideband codec as recommended by Global IP Sound. GIPS iPCM is a wideband codec that is extremely robust to packet loss. It provides virtually no degradation of quality at 30 percent packet loss and conversation is still possible at 50 percent packet loss.

We tested the sound quality of the GIPS VoiceEngine by making some test VoIP calls. For our first test, we added a contact within the iPAQ�s contacts screen, namely the tech-support contact at Global IP Sound. We should mention that all that was really required in the contact info was an e-mail address. Next, we connected stereo headphones connected to the iPAQ to minimize feedback into the iPAQ�s microphone. Then, within the Pocket Presence application, we entered the correct IP address of the locator server that provides the translation between the contact�s e-mail address and the contact�s IP address.

After highlighting the contact on the iPAQ, we simply clicked the green phone icon to initiate a VoIP call connected via the Internet. We were connected with the tech-support contact and we had a five-minute conversation with each other from our iPAQ to his. The voice quality was amazingly good even across the Internet.

Next, we did some internal testing using the Pocket Presence client installed both on a desktop PC as well as the iPaq. We should mention that both the iPaq and PC graphical user interfaces were nearly identical. We also utilized Global IP Sound�s server piece of software installed on a second desktop PC, which handled the translation of an e-mail address to an IP address. The desktop PC was connected to our 10/100BaseT Switched LAN while the iPaq was connected to a Linksys WAP-11 WiFi Access Point that was then connected to our switched LAN.

We then configured the first desktop PC to point to the IP address of the second desktop PC acting as the e-mail address-to-IP address translator -- a pseudo SIP or H.323 registrar, so to speak. We performed the same procedure on the iPaq, pointing it to the second desktop�s IP address. We placed the PC�s microphone in front of the PC speakers to record the time when the iPaq transmitted some sound to the PC speakers, which was used in our latency calculations. To measure the latency we played a quick sound into both the iPaq�s and the PC�s microphones. We should mention that we were running the CoolEdit application running on the PC to record and time-stamp the short sound we played. The sound played into the iPaq microphone then traveled across our WiFi network (via the IP protocol) to our 10/100BaseT switched LAN, into the desktop PC, and finally was played out the PC�s speakers. The microphone in front of the speaker�s picked up the sound event, which was also recorded and time-stamped by the CoolEdit application.

We should point out that there is some extra latency due to the time it takes for the sound card to process the sound and play it back of the speakers. Typically between 5-15ms for a PCI sound card and 20-25ms for a DMA-based sound card. For our tests, we used a PCI sound card. Since the latency is minimal, we did not subtract this latency from the final calculations.

Measuring The Latency

By examining the waveforms in CoolEdit, we were able to see the initial sound event and the subsequent sound event, which essentially was the one-way trip for the initial sound (iPaq) to travel to the other end (PC). We simply subtracted the time difference between the two to achieve our latency measurement or simply highlighted the area on the screen and let CoolEdit automatically calculate the latency. We highlighted a section of the waveform from the start of the first sound event to the start of the second time event, which gives us the time length or latency. The calculation of 171ms is shown in the lower right-hand corned in the �Length� field.

We performed several tests using all of the various codecs with no packet loss. Then, using Shunra�s Cloud 4.0, we induced 25 percent and 50 percent packet loss to see how that would affect the latency (Table 2).

We noticed an interesting thing when we examined the calculations in the tables. When testing PC-to-iPaq latency, at 0 percent packet loss, the latency was 204ms, and then as expected, increased a bit to 392ms when we set the packet loss to 25 percent. However, unexpectedly, when we increased the packet loss to 50 percent, the latency went down to 211ms. Similarly, when testing iPaq-to-PC latency, we calculated 344ms at 0 percent packet loss, but the latency decreased to 259ms at 25 percent packet loss and decreased even further to 230ms at 50 percent packet loss. Seems a bit counter-intuitive, however, we believe this to be due to the design of Global IP Sound�s codec, perhaps having to do something with packet re-transmissions or adjusting of the jitter buffer.

Voice Quality

We listed our perceptions of the voice quality using Global IP Sound�s proprietary codec, which was designed to handle high packet loss (Table 2). As expected, as 0 percent packet loss, the voice quality was excellent. Unexpectedly, we were very impressed with the quality at 25 percent packet loss and even more impressed that the voice quality was still good at 50 percent packet loss. We expected words to be cut off, or at least a few syllables, but actually this did not occur. Although, we did notice that at 50 percent packet loss, eventually the VoIP call would often be terminated, which then required that we reconnect the VoIP call. So we do wish that their VoIP soft-clients would automatically reconnect when the call is dropped due to packet loss.

For comparison to Global IP Sound�s special codec, we also tested the voice quality using the G.711 and Enhanced G.711 codec with 25 percent and 50 percent packet loss. For both codecs, we found that the voice was choppy, words were missed, and it sounded �robotic,� especially when there was a stream of packet loss. It was quite clear that Global IP Sound�s proprietary codec was much better at handling packet loss than the other codecs with virtually no degradation in voice quality.

Conclusion

The iPCM-wb NetEQ codec has advantages over other codecs, including the G.729 codec, which is a popular codec often used in narrowband applications. For instance, the G.729 codec is only 8 kbits/s and can only handle less than 10 percent packet loss before the conversation becomes unintelligible. The GIPS iPCM-wb codec can operate at a higher bit rate, 80 kbits/s and can handle 50 percent packet loss while still maintaining acceptable, if not �good� voice quality. In addition, instead of using the usual sampling 8 kHz frequency typically used for telephony band products, iPCM-wb uses 16 kHz for wide-band speech coding, producing much more natural, comfortable, and intelligible speech. Wideband codecs cannot be used when communicating with the PSTN where only narrowband (8 kHz sampling) is supported. So in essence, the iPCM-wb codec can actually sound better than a traditional PSTN call.

Using Global IP Sound�s codec along with an application such as Pocket Presence installed on a network-connected PocketPC will allow anyone to make phone calls across the Internet bypassing expensive cell networks. In areas where 802.11b coverage is wide, such as some city districts, college campuses, etc., both making and receiving phone calls can be easily done. Overall, TMC Labs was very impressed with how Global IP Sound�s codec handles packet loss while still maintaining voice quality. Both Global IP Sound�s codec and the Pocket Presence application are certainly worth a look if you are looking for a VoIP solution.

[ Return To The February 2003 Table Of Contents ]



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