I tried a few high quality VoIP providers, including two of which I have had an account for long and run my asterisk server on.
They offer flawless quality in VoIP when I do phone-2phone bridging (my asterisk server initiates 2 phone calls and links them together, removing the internet latency factor all together).
However, Ive decided to use my TyTN for VoIP as I might give up my ground line & give up unlimited incoming call on my cellphone plan.
After testing my TyTN, I'm very unsatisfied. There is a slight stuttering and it gets worse when both sides are sending noise (if only 1 person speaks and the other listen, it's fine with a bit of stuttering, but if the listening person starts speaking, then the stuttering will get pretty bad).
I got QOS on my router for SIP VoIP service and I've tested the quality of calls when my router had no bandwidth stress AND by calling information lines (automated speaking) to remove as many variables as possible.
The matter is, the stuttering remains.
Yet I see a lot of people saying they use their TyTN on VoIP for everyday use.
Do you suffer from stuttering phone calls? Or do you use a magical VoIP provider? or do you have a special router/asterisk/WM6 config?
thnx!
EDIT - This post was originally about some other problem which I fixed. SO if you read about the speakerphone.. forget about it I fixed it by soft resetting, im stupid.
Related
Frustrating stuff and i'm not sure if it's the phone or the network.
We bought 35 HTC TyTn's recently on the vodafone network and everyone's complaining about dropped calls and having to make multiple attempts before connecting.
We have done a test that confirms both mobiles have 100% signal and neither phones were being used and still we encountered that frustrating "BEEP BEEP".
Any ideas?
Also, have any of you noticed that the there's a bit of delay on the keypad? Any way to resolve this as it's a little annoying too!
HELPPPP!!
J.
Does the "beep beep" sound more like a soft musical double tone than a beep? I've been noticing this myself lately when trying to make outbound calls but I thought it was something to do with Cingular's network. Are you running the stock TyTN ROM?
attempted help
I was having this quite regularly on my XDA2 with O2.
I recently hard reset the device, and didnt restore my backup. Since then problem solved.
Could it be other software interfering?
Dan
If it's the monotonal "beep-beep-beep" sound, it usually means the network is busy- I get this in London quite a lot when on a UMTS connection, but seldom get it out "in the sticks" and almost never when on a standard GSM connection. I guess it depends how busy your local tower is. If the tower's congested, you'll get this tone even if you have a full signal.
As is goes, we had this problem at our workplace where there are around 400 people all using Voda (obviously not all at the same time all the time!!). Voda UK kindly installed a tower upgrade for us. Maybe worth a call to CS if you're a fairly big corporate customer.
As a slight aside, there are a few levels of priority assigned to SIM cards- if someone with a higher priority makes a call when the tower is full, someone with a lower priority gets chucked off. My understanding is that on most UK networks, there are 3 or more priority levels (with emergency services getting the top priority, understandably, big corporates getting priority 2, and everyone else getting priority 3; some networks assign the lowest priority to PAYG customers).
I sometimes get a delay on the keyboard, but usually because I have a load of apps running in the background- some cause this problem even after they've been closed for some reason (stand up Mr. TomTom 6). I find a soft reset solves it.
That wouldnt although explain why my reset has solved the problem.... i was getting that issue daily, now havent had it in 3 days.
Strange.........
Dan
Hi guys,
I want to use my HTC Wizard to call out on the "normal" or DSL phone line when I am at home. This beacause all phone calls to all national non mobile numbers are free.
My PC is connected to the DSL Modem (ZYXEL P-2602HW-D1A) and with a modem connected to the phone out put of the modem and to the normal analog line.
Is there any one who has experience with this?
MartindH said:
Hi guys,
I want to use my HTC Wizard to call out on the "normal" or DSL phone line when I am at home. This beacause all phone calls to all national non mobile numbers are free.
My PC is connected to the DSL Modem (ZYXEL P-2602HW-D1A) and with a modem connected to the phone out put of the modem and to the normal analog line.
Is there any one who has experience with this?
Click to expand...
Click to collapse
The answer is likely to not be as easy as you expect, although it may not be that hard.
Your modem may or may not work, when you do voice over a modem the modem needs to know this and not demand a carrier tone and other things that signal a data connection. Some modems are known to work in this capacity others are known to work very poorly (lots of echo) and others are known to not work at all. The majority of modems are not known whether or not they will work at all.
In short the easiest way to accomplish this task is to send data from your phone VoIP to your PC. There are free clients out there such as sjphone from sjlabs.com. You will likely want a headset on your phone as most dont use the same speaker as a regular phone call but instead the speakerphone and echo cancelation doesnt work well (the remote side will hear echo without headphones).
Now that your phone is taken care of you need something on the other side. Here you have choices. If you have a compatible modem you can use that as an FXO card with software like asterisk.org and soon freeswitch.org. If you do not have a compatible card, or do not wish to run VoIP software on your PC you can get an ATA that has an FXO port and lets you route calls to/from it. Grandstream.com has some, the HT486 comes to mind. I believe the linksys pap2 will also do this. Ebay may be your friend in locating a fairly cheap one, although they arent that expensive - and you are doing this to save money so depending on the number of calls you make it may pay for itself soon
Once you have this set up, you can actually choose to call people via your mobile plan or landline or even an internet based telephone company. Depending on how well you configure everything, you could in theory have it use all of those things, and you can even route calls from those services to your phone (ie get phone numers all over the world and answer them on your mda when you have internet).
All your mda needs is wifi/usb/gprs. And for those providers that block VoIP on gprs shame on you (and they generally dont block vpn traffic or even know what the contents of that are
Port restricted Cone NAT
Thanks for your support.
The WIFI way with a direct connetion to my Modem will do for now, but I have got the following problem there.
The error that is displayed is the following:
NAT/Firewall: Port Restricted Cone NAT
The settings which I have entered are the same as in my Modem:
Zyxel: P-2602HW-D1A
Provider Tiscali
Anybody who knows how to solve this or who has experience with VOIP provided by Tiscali or other ISP using your HTC Wizard
hi,
i can't seem to get it working. people can hear me, but i can't hear them
Anyone got this working?
Thx
it should work, i have not tested it but it should work, the files and registry entryes are taken from a hermes rom for which i am sure that it work and it is the same core and build as ours
i was able to launch the setup earlier... and decided to do it later on..
but when i tried after 2 days ... it just crashed...
did uninstall... softrest... install.. routine few times... and gave up.
I successfully connected to my Hungarian Voip provider (megafone) but no sound at all. Dialing is okay, but I cannot hear it ringing out, cannot hear the other party, they can't hear me. However the connection is there and the connected seconds are charged.
GSM phone, Skype and Fring are working well, only the Ms Voip has problems.
My settings look like this. The 'Dial plan' tab is empty. Is this normal?
Also noted that it didn't terminate the call normally as I pressed the red button, it kept charging on for 15 minutes of talk...
well now i think that something is missing, maybe audio encoders and decoders on which VoIP is dependant, i will look into this and will keep you posted
My tip is also codec-related.
Is it normal that via ActiveSync the voip says No Service? With WiFi it says Available, I think it should also with ActiveSync.
are there any news?
I don't have an issue with VOIP working on my SX66. In fact, I was surprised by how easy it was to get going.
-Chris
You were lucky that your provider uses the same codec as this WM6 build does.
Since I don't use something like Vonage, or Comcast VOIP, I just went out and found a VOIP provider online. VOIPCheap.com works great with this build.
-Chris
Many of us prefer a local VoIP provider though.
VoIP Incoming RTP
Some additional info on the problem with receiving audio. (same is on my phone). I checked which codec this SIP stack negotiates.
Apparently, out of all available, MS VoIP settles either on PCMA or PCMU depending on the other end availability, and refuses other most common ones.
Well, it encodes and sends RTP which I receive on other non wm6 phones and I am sure it receives RTP, and may even decode it, but by the look of it, not a squick is sent to the hardware/speaker.
Cheers
===
Sorted by installing Voip support from (ftp://xda:[email protected]/WM6VoIP.CAB)
and installing on top of the existing (from voip.cab - can't remember the source).
Although the dialplan tab is still blank (requires ipdialplan.xml, I guess) but the SIP settings are remain intact after this 'patch'.
====
I've just tested the MS voip with 3 different VOIP providers, and I also have the "sound" problem with it.
When I use an other VOIP application (like SJphone) all is working fine, so it must be a codec problem that is missing.
I just looked at VOIPCheap.com website. It's sounds interesting. Never tried VOIP from BA. Can someone tell me what i need to do to use VOIPCheap.com on BA to make VOID calls apart from creating an account with VOIPCheap.com? Thanks guys.
Never Mind. Please ignore my last msg. I think i got the answer on there website in FAqs. I'll create the account and see how good it is.
Anyone have any good/bad feedback for VOIPCheap.com service?
Thanks.
There are a lot of threads for VoIP problems in the forums many issues are solved there, i suggest al of the VoIP users to search and read there. Also i understood that difrend providers use difrend encoding-decoding so thats why it works for some and not for everybody
did someone find a fix for the muted audio issue?
i've read many threads about many voip issues on the other forums available here, but it seems like the small amount of ppl who had this problem was left quite alone..
I'm trying to place calls via my asterisk 1.4 box.
thanks
Hello guys! I have been trying to make internet calls with my new HTC HD2, and so far I have not managed to make it work properly.
Actually after testing PortSip I almost got it, but the problem I have with this software is that my voice is not being heard at all/clear enough. I can hear the person I'm calling very well, without delays, but they can not hear me. I tried using BT headset, normal headset, and without headsets for the same result, not being heard.
Could it be that the microphone is set too low? is there any way to increase it?
Any help would be appreciated.
Thnx.
I use PortSip without any such problem....
Try fring as an alternate solution...may work for you...
Are you connecting to your own VOIP server? If so, make sure that the correct ports are open.
I had the same problem with portsip. I find Agephone to work nicely, only problem with it not being able to direct sound to the earpiece so you will need some software that does that instead.
After a lot of testing, I'm still in the same situation. I'd be great to make voip calls to work. I am certain there's some settings in windows mobile (codecs maybe) that my voip provider don't like, and therefore calls drop.
My voip provider is Saunalahti (Finland) which actually rents the gsm network from Elisa.
I am trying making calls from two different voip accounts, calling other voip numbers and normal gsm numbers. I am using one HD2, one TYTN and a Softphone that Saunalahti offers.
The only way that the calls work is calling from the softphone in the PC, no problem there. I have audio both ways and calls do not drop.
If I call from:
wm to PC, call drops (lasts 2 sec)
wm to wm, call drops (lasts 2 sec)
wm to gsm, connects but the other person can not hear me
One wm is using 3G and the other is using WIFI. port 5060 is open in the registry.
Like a said earlier, I have tried different programs, but it's the same scenario.
Any idea on what the problem actually is? thnx i nadcance.
Hi guys,
I am interested in VoIP calling with my HD2. I don't know much about this, and I would like to be able to use it when I travel to Mexico next month. I know that Blackberry's have UMA technology or soemthing like that to make WiFi calling for free back to the states. Will this PortGo/PortSip program do the same thing if I use the free version, or buy the Professional version?
האם אפשר לקבל עזרה בעיברית להפעלת voip
gorelshv said:
i had the same problem with portsip. I find agephone to work nicely, only problem with it not being able to direct sound to the earpiece so you will need some software that does that instead.
Click to expand...
Click to collapse
האם אפשר לקבל עזרה בשפת הקודש?
So, I've decided to give WiFi calling a spin (despite having a solid 4 bars of sweet, sweet HSPA+ sitting at my desk). I'd like to share my findings with you guys.
Firstly, I've looked into the bandwidth used by WiFi calling, by watching my router's information page.
Here is a screenshot of an active call
The very first section is me tapping on the mic, while waiting on hold with HTC (that's a whole 'nother story). The second part, silently listening to their boring hold music/we appreciate your call as a valued customer crap, the third part, a two-way, engaged conversation, and finally, me giving her my name, ticket number, etc, while she silently listened.
By the way, despite the green line being stated as "in", it is actually the outbound (upstream) bandwidth from my WAN interface. It's reporting as "in" because the data is flowing out of the WLAN chip on my phone, in to the WLAN interface (wl0) on the router, then going through the bridge (br0), and out the WAN port (vlan2) on my router. And, in effect, the red line would be incoming (downstream) data on my WAN, and it is flowing out of wl0 into my phone. Hope that makes sense. Also, my computer that I was viewing this info from did not fudge the results at all, I am connected via wl1 (the 5 GHz interface), and my G2 was the only client on the 2.4 GHz interface.
During this call, there was also some other traffic flowing in/out of the WAN interface (traffic from my PC, broadcast messages, and what not). Nothing extreme. The maximum bandwidth used at any point in the call was about 80kbps (kilobits per socond) in either direction. I am going to go out on a limb here and guess latency is more important to call quality than available bandwidth.
I do not have packet-level QoS enabled in my router, however, I do have WMM/WME (Wireless Multimedia Extensions) enabled, with DD-WRT's default settings. I am unsure if Kineto's UMA app tags the frames with the necessary information for it to be prioritized by WMM/WME. I guess I will look into that later.
From a subjective point of view, the woman I was speaking with sounded crystal clear, quite possibly better than a typical UMTS/GSM based call. I also asked her how I sounded on her end, and her response was "absolutely gorgeous". Then, we got back to the actual conversation, and about 30 seconds later, the call dropped. Hmph. The call drop log in SETTINGS --> ADVANCED SETTINGS --> CALL DROP LOG is as follows:
Code:
CD-12 ISP Problems
2010:11:04:11:06:18:336
Perhaps we can find reference to all of the error codes somewhere, and find out exactly what went wrong?
Well, that just about sums up my findings and information for now. I will be looking deeper into this, and how we can make it better, and more stable. I will also try to keep this post updated with my findings, as well as those of others, so you do not have to read through many pages to find all of the information.
thanks for the data. Glad to hear that wifi calling only sips a little. I'd like to see some battery life info though.
Cool!
Yeah latency is allways goign to be key - in a few aspects:
High latency is a problem: this would be due to you saying something and it not getting to the other end in time (this is why wifi calling over 3/4g is a bad idea).
If it takes too long it's discarded (choppyness) as we can't retransmit parts of a conversation over other parts that are already occuring in real time.
Drops: If a packet drops it's gone, it's not retransmitted - this is UDP, the reason being, again if you say something and it doesn't get there - there is no reason to retransmit it as you are already on to the next word in your sentance.
Of course in a TCPIP situation, for example accessing a website - if your initial request drops it's just retransmitted, several times if neccessary - the only thing you notice is waiting an extra half of a second for a page to load.
Jitter:is the variation in latency. If your latency is jumping all over from 40ms to 120+ms and back again then you will also experience issues. If there is slightly higher but consistent latency on a connection then this is likley better then a high varation in jitter.
I wonder what all codecs they're all using to transmit ... if it's straight voip or what?
khaosxiii said:
I wonder what all codecs they're all using to transmit ... if it's straight voip or what?
Click to expand...
Click to collapse
It isn't VoIP.
It is GSM over IP. GAN/UMA/WiFi Calling takes the normal GSM packets and encapsulates them in IP packets and send them thru an encrypted tunnel back to the carrier's (TMO) network. Once at the carrier network, the GSM packets are stripped out of the IP packets and handled like any other GSM call.
If you are worryed about latency then turn QOS on, on your router and set your phone to have a static ip when you are at home. Thats what i did for our VOIP router at my house. Should work the same with your phone you would think