ageet Forums

We use VoIP on a regular basis for our telephone communication needs. Our current setup uses both a LinkSYS adapter for standard phone hookup and X-Lite softphone on XP PCs. Our goal is to use my cell/pda (HTC X7510) as a 3rd alternative when I am travelling for business.

We tried "fring". It works but the sound is horrible.
We tried "destr0_VOIP_WM61___SIP_Config_1.1.CAB" from a forum on the web . We were pleased to see it totally integrated with the phone (no need to launch another application) but unfortunately the sound was even worst than "fring". And since this application was not backed-up by a true company we had no support at all.

So we were pleased to see "AGEphone mobile", supported by what seems to be a true company behind it so we tried it. It worked immediately so we purchased the license right away.

After using it for a couple of times, a few problems started to show.

1) The sound is actually quite good on some occasions but for some reason it is inconsistent. There seems to be crackling sounds on some calls or in several portions of a conversation. While on other occasions, the conversation is perfect or very near. Our settings related to sound are as follows:

- Audio Packet Sound: 20
- Audio Buffers (playback/record): 120 or 180 ms (current 2 values under test)

We tried increasing the audio buffers to eliminate the crackling sound but it changes nothing or very little for the quality but adds more delay which is our problem no.2 below.

2) The delay between our voice and the other party is longer than acceptable (when we compare with our other VoIP solutions above).

3) When we access our voicemail, we have 2 problems. We don't hear or we hear badly the voice giving us instructions (crackling sound or no voice at all) to get our messages. And when we are instructed (after a while we hear the instruction voice) to press the keypad it seems to work but it doesn't sound from the speakers.

The last problem above we could live with by taking our messages later from another system. But due to the first 2 problems, it affects the quality of our conversations with our clients and therefore makes this software unusable for us at the moment. It is really too bad since it is the best we tried so far and we really believe we are not very far from a true VoIP mobile solution.

Please help.

Hello joker and welcome to our forums! It's good that you gave AGEphone Mobile a thorough test and I hope that I can explain some of the issues that you have listed. Let me start with a little question: Are you using wireless LAN, a 3G network or both when you get inconsistent call quality (1)? If you are running on a mobile network, please keep in mind that with todays equipment many factors come into play that can make the difference between good and really bad call quality. These are:

- Cell congestion: The more 3G devices (phones, dongles) you see out there the more likely it becomes that there are network cells that are running at the edge of their capacity.
- Network congestion: While the current cell might be unstrained, many mobile operators didn't invest enough into their backbones and bendwidth may be limited again.
- Weak signal: It has been found that on anything but the strongest signals bandwidth decreases sharply in favor of devices that have a better signal.
- Your VoIP provider might be using shady routes if the tariff is very cheap. This goes especially for Betamax providers. Which VoIP provider are you using?
- The PSTN gateway could have congestion issues as well.
(-Using BT (headsets) influences the the WLAN and 3G radio effectivity in some cases. Sometimes BT headphones crackle when the call quality itself is perfectly fine.)

Of course the callee could suffer from the same conditions and thus it is nearly impossible to guarantee a constant and good quality while using VoIP over 3G. Please note that I am not even talking about GPRS and EDGE here as both standards are not suited for VoIP. Even if their theoretical bandwidth is sufficient they introduce 500 ms of delay from the base station to the mobile device alone.

If you are using a 3G network, 120 ms is the bare minimum for the buffer. 180 ms should do the job just fine. More is rarely needed as it introduces too much delay (2). Speaking of which: AGEphone doesn't introduce any unnecessary delay by itself. The value consists of the buffers, the distance to the base station (100 ms with 3G) as well as the distance between you, your VoIP provider and the called party. The only part of this formula where you (and other software) can save is the buffers. The bad quality of the other products might in part be a result of the buffers being set too low. When it comes to the audio packet size, a value of 40 seems to be the best choice for most mobile networks. Experimenting with this setting can improve sound quality a lot if you experience periodic audio skipping.

The last issue you mentioned is the voicemail (3). The sound quality should not be any different from a normal call here, but please let me know which provider you are calling. As for the DTMF tones, AGEphone Mobile supports two methods of sending them (take a look at page 3 of the provider options), but neither of those plays the tone while dialing it. I confirmed that this is also the behavior of many normal phones and we intend to leave it like as the feature itself works flawlessly and you can make your selections.

All of this means in short that we did all we could to guarantee a good quality on AGEphone's side, but there are many factors that we simply can't influence and hence it is difficult to recommend VoIPo3G for business situations that rely on absolutely perfect call quality.

Thanks for your quick reply.

- Are you using wireless LAN, a 3G network or both when you get inconsistent call quality (1)?
Wireless LAN. I only use VoIP on WiFi networks at the office or in hotels. On the road I use the cell only.

Today we compared both the laptop and the PDA by putting them side by side on the same WiFi network at the office. The PDA was running AgePhone Mobile (your latest version 10102008) and the laptop was running X-Lite. The laptop communication is perfect and reliable all the time. The PDA communication was again inconsistent meaning crackling/interrupted sounds showed up here and there no matter what settings we used. But between these interrupted sounds the sound is perfect. We have several VoIP accounts with different settings for our testing. The results are never the same. On some occasions, the communication is good but unfortunately it's not always the case so we cannot rely on it as we don't know how it will turn out. So the issue is clearly either the PDA itself (HTC X7510) or the software. I could only confirm that it is the software if only I could find another software solution but from my experience so far you are the best we've found (as mentioned in my previous post the quality would be good enough for business if only the reliability was there).


- The last issue you mentioned is the voicemail (3). The sound quality should not be any different from a normal call here, but please let me know which provider you are calling.
Indeed but unfortunately at the start of the voice mail message instructions the first words are truncated (we only hear the last word(s) for some reason). Then it asks to press keypad numbers for instructions and they are not received all the time. The VoIP provider is 'Acanac.com'. We use them both in our computers and LinkSys adapters.

As for the DTMF tones, AGEphone Mobile supports two methods of sending them (take a look at page 3 of the provider options), but neither of those plays the tone while dialing it. I confirmed that this is also the behavior of many normal phones and we intend to leave it like as the feature itself works flawlessly and you can make your selections.

I will try both options and see if this explains why on some occasions it works and on other it doesn't (maybe we played with the settings).

Thanking you in advance.

Are you using a Bluetooth or wired headset with your HTC X7510? Did you try audio packet sizes other than 20 ms (even if you are on WiFi it might help)?

Wired headset only as our goal at the moment is to have a bare minimum system running reliably (no other applications running, AC connected, no other communication sessions open - cell phone and bluetooth are off, ...).

Regarding the audio packet sizes, we tried all values, from 10 ms up to 50 ms. we are currently set at 30 ms.

Yesterday we tried "Pocket Talk Business Edition" from NCH Swift Sound (sorry for mentioning one of your competitors but I think the info could be good for the purpose of this problem) and we found they have one interesting feature which are Audio and Network Setup Wizards for setting up the application properly according to the device. During the Network analysis, more specifically during uPNP NAT analysis, the following message pops-up:

"Manual Router Setup Required.
A 'symmetric' NAT or router has been detected between this computer and the internet. This means audio cannot be routed to this computer automatically. You (or your Network Administrator) are strongly advised to manually setup RTP/UDP redirection at the router or NAT otherwise audio may be delayed, low quality or broken."

Both the "delay" and "broken" audio are in fact the problems we are experiencing with your (and this) application on our X7510 phone.

Last week before testing Ageet Soft I also try "Pocket Talk Business Edition" from NCH but I have the same problem than you joker (same pop up message). Its strange but if you read my thread in this section I have the same problem than you with Ageet using an HTC Touch Diamond. Both of us have the "broken" audio problem and are in fact the problems we are experiencing using Ageet application, the strange thing is why I dont have any problems using sjphone soft... may be its not a NAT problem and NCH have a bug with this pop-up message and Ageet have another problem...

Thank you both for the input and I think that joker might be on to something here. Usually, with "Symmetric NAT" things either work or don't work at all. If your calls connect there shouldn't be any adverse effects on quality or latency. But to make really sure, could you try to either enable UPnP on your router or alternatively forward the according SIP and RTP ports manually? If you can't change these settings, testing from another hotspot might also solve the problem. Once the "Symmetric NAT" message is gone, please test the audio quality again and let me know if AGEphone Mobile works more reliably for you.

I'm just another user here...
but I use VOIP over wifi a lot (HTC Touch)
from my humble experience if the Internet connection works well there's not better sound around than Agephone's...

ppatrick

Thanks a lot for the feedback ppatrick which is reassuring to hear! We usually don't receive negative reports about the call quality and so I am still very interested in how this case has worked out for joker and Flashhh.

Sorry for taking so long to post back but unfortunately none of my "debugging" experiences made any difference. I tried to manually forward the ports and I tried in several hotspots while on the road; still the same.

So I installed another software, SJPhone, and it worked right away by properly resolving the port/Nat issue (if there was one in the first place) and connecting to all my VOIP accounts. The delay is sometimes perfect and sometimes noticeable; I know it does not depends on the router/location since I've experienced both scenarios at the same location. The sound quality is quite good on my end (not perfect at all times but good enough at all times for a clear conversation) but very good or perfect on the other end (my customers) which is what is important to me. Therefore I have been using it with my clients for a couple of weeks now and I can say I am confident enough to use it on a regular basis.

This really points out to a problem in your software and unfortunately I don't have time to test any further on your behalf. It is unfortunate that you don't have enough manpower to properly develop your software and need to rely on your customers to test it for you. But what is more unfortunate is that I need to rely on a 5 year old software for my everyday needs and put my Ageet license aside.

I will be installing updates from your website on a regular basis and see if one day you will have a properly running software.

We are a small company, but the difficulty with your problem lies elsewhere: We cannot reproduce it on our end and therefore have to rely on your input which we take very seriously. That being said, you and Flashhh! were the only ones to ever report problems with the call quality. I am using AGEphone Mobile on a daily basis and with different providers myself and never ran into any problems. So until we get more reports that help us to pinpoint an error in AGEphone Mobile 2 there is nothing that our programmers can do about this.

Michael wrote:...That being said, you and Flashhh! were the only ones to ever report problems with the call quality.

Dear Michael,

I am in a similar situation as described in this thread. I am looking for a client to use with my companys' Asterisk server. I have come across the different software products mentioned in this thread, but your product seems promising, however I do experience the same problems.

Client: HTC Hermes, WM6.1
Server: Asterisk 1.6 (configuration can be adjusted to the needs of agephone)
Connected via WLAN/LAN (about 7 hops away, sufficient bandwidth, no congestion or other issues)

I have incoming&outgoing sound, but it is not really stable. I have tried different buffer settings and packet sizes, but I do not really like the idea of network dependant tweaking, as the primary purpose of the mobile device is to use it on the go in different WiFi locations.

I guess the problem me and the other posters are having is the difficulty to nail down the culprit for these lags and cracking noises. To rule out other problems, are there any special asterisk configuration aspects that you would recommed? Anything we might have misconfigured on WinMobile-side? Or can you suggest a certain provider and settings for testing, in order to rule out other mistakes? Or is it may be a hardware issue?

Thanks,
martin

Hello Martin and welcome to the forums! You are right: This issue is really hard to determine and we might be talking about hardware issues here. Are you using the internal or external speaker when experiencing the lags and cracking noises?

martin wrote:

Michael wrote:...That being said, you and Flashhh! were the only ones to ever report problems with the call quality.

Dear Michael,

I am in a similar situation as described in this thread. I am looking for a client to use with my companys' Asterisk server. I have come across the different software products mentioned in this thread, but your product seems promising, however I do experience the same problems.

I just read this. Michael, when I read Jocker problems I sent a PM to him and suggest him to test with Sjphone to discard possible voip providers problems and Jocker confirms that the problems are with AgePhone Mobile, now martin have similar problems, and also users of xda-developers.com. I have and HTC Touch Diamond and a new HTC Touch HD, my brother have another HTC Diamond with different ROM than mine and all the 3 devices have the same problems with AgePhone Mobile soft and not with Sjphone. You should test with this model devices and you will see what kind of problem we are talking about, you will confirm that we are not making nothing wrong. Not seeing this soft problem is an error. Im making an effort trying to help you and your team to solve this problems because I like that Ageet are trying to make a good Sip mobile soft that replace the old Sjphone and I see that you are going in the correct way and your soft seems promising. I will switch from SJPhone and buy AGEPhone licenses when this problems were solve. If you need something from me ask and I will try to help you...

Hi Guys
Same situation here.
Love the Ageet client & hoping we can recommend it to our own clients.
Using on HTC Touch HD & really hoping audio quality can be addressed.
Previously used the NCH client without issues, but the Ageet is definitley the preferred product if the audio issues can be resolved.

CD (UK).