Problem of VoIP

I would like to develop a java VOIP program with the uses of JMF.
But i have no idea of how to write the program like this.
Can any one tell me?
Or can any one give me the source , so that i know how to develop such a program.

True, but you will get notification by IOException.
If you use broadcasting (UDP), the receiver is no
longer listening, will you get IOException? No.
What I meant with the guarantee here is you can
design your code with a way to sure the package is
delivered, or else exception handling can be executed
to minimalize the failure.Yes you will probably receive an IOException eventually. But
after how log will you receive it? It could take a couple of
second/mins if the receiver just drops of the net as you would
have to wait for the TCP timeout to occur. (basically you have to
wait for whatever period of time this is set to).
The main reason UDP is chosen for these kinds of applications is that
the send/ack/ackack protocol that
tcp uses produces high latencies (three network traversals) and for the
reasons explianed above is not guarenteed to produce prompt error
notification. If this is the situation then why bother with the protocol
at all. Just use UDP (it does not have the same latencey issues) so can
be faster and, like TCP, also does not provide prompt error
notifications. In fact to receive any kind of error you will need your
client to send "I'm still here" messages to the server.
Packets are not guaranteed to be received in order.Several packets can be sent but the next one will not
be delivered from the TCP stack until that packet has
been verified. They will be delivered in order. This is very true. The stack should reassemble the packets into
an ordered datastream. However very large latencies can be introduced
if one of the packets (from the middle of the stream is missing). The
stack will then not provide any of the later packets in the sequence
until the missing packet turns up or can be re-requested. While this
is happening your client is starving for want of data. Using UDP it is
often possible to just skip the missing packet
(causes a slight glitch in the sound you are hearing) rather then the
pause in the sound you are likely to receive via TCP connections.
Basiclly, when using the internet, very few guarentees can be made about
any transmission protocol. Niether UDP or TCP is designed for fault
tolerant operation under strict time constraints over such an
unpredictable medium. UDP guarentees very little. TCP guarentees a lot.
The guarentees TCP makes are of little use in such an application while
the overhead created to ensure those guarentees is expensive. That is the
reason UDP is often chosen over TCP for applications with strict time
constraints.
matfud

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i see the little wifi-connected icon along with the voip icon. and i can make outgoing calls ok. and for awhile i get incoming via voip, but after a period of time, incoming voip doesn't work anymore.
anyone else with this problem?

Then I'm afraid I don't have a solution. This was a common fault with early N95 firmware versions - when the phone went into standby mode it would shut down the WLAN connection to save power regardless of whether or not it was actually in service.
Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you!

E52:the problem in voip

I have a problem.My friend has E66 and i have E52.And I found E66's Voip can support both wifi and 3G internet.But E52's voip just can support wifi.And i need my E52 voip can support 3G internet.So i survey the voip version of E52 and E66.
http://www.forum.nokia.com/Technology_Topics/Mobile_Technologies/VoIP/Nokia_VoIP_Framework/VoIP_supp...
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Solved!
Go to Solution.

To solve this go to "Advanced VoIP settings" menue.
Select "VoIP services" and choose your configured service.
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Go to the entry "AWCDMA" and set it to "on".
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WP8 WASAPI Low Volume Audio Recording Problem on VOIP App

Hello,
I am working on a VOIP app and WASAPI is used for the app. I am using WASAPI in SHARED_MODE. Captured audio frames are turned out to be in low volume. For some devices, the audio volume is almost silent (Specially for Lumia 730).
From the VOIP app documentation, i found out that only ISimpleAudioVolume is supported for volume control for Windows Phone. (Ref: https://msdn.microsoft.com/en-us/library/windows/apps/jj715884(v=vs.105).aspx). But ISimpleAudioVolume::GetMasterVolume(float
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Haider

Hi haiderBd,
In fact, Only ISimpleAudioVolume API is supported for Windows Phone 8. IAudioStreamVolume or IChannelAudioVolume is not supported for Windows Phone 8. Am i right or i have missed something else?
-> Yes, you are correct, if you read carefully about the Requirements section, you should be able to find that ISimpleAudioVolume supported on Windows Store App, but it also mentioned does not supported on Windows Phone 8, which means the API is acceptable
for Windows Store App for instance PC and tablet.
Again, IChannelAudioVolume only support on Windows Desktop, which means the api can be only used in desktop development for instance WPF, winform and etc...
What could be the reason for low volume recording? Is there any way to increase the volume?
-> I'm not that familiar with WASAPI, perhaps related with Share Mode? Try with exclusive mode to see if listening to one audio endpoint device while recording.
I will also try to find some experts to help you with this question.
--James
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Annoying sound problems in voip programs.

Hello,
I have a x-fi sb soundcard.
And I have problem: Whenever I am in Teamspeak or another voice program and someone says something, they'll hear a strange noise coming from me, which is exactly as long as they talk and with the same rhythm and really annoying for them. I'm using a headset. I can also disable my microphone and they'll still hear themselves (well not really, a loud ugly noise). If I try another headset the same problem occurs.
Here is a screenshot of my sound settings <a rel="nofollow" href="http://www.abload.de/img/aufzeichneny40z.jpg"]http://www.abload.de/img/aufzeichneny40z.jpg[/url]
Would be awesome if you can help me, this is bothering me for months.

- Can you look in your control panel under sound, and if you multiple tabs, do you have one for Audio Enhancements? If so, can you uncheck the box, reboot your machine, and try your team? speak again? Also, if you have exactly four gigs of RAM in your system, you might need to limit yourself to just under 4 gigs. You can do this by going to start, run......and then type msconfig. There you will find a tab that says.........boot, and then to advanced options, there under maximum memory, you can type 3980. This will limit your RAM to just under 4 gigs. Then you do not have to remove any sticks of memory, and your microphone should work.
Here is a screen shot........
<img border="0" src="http://i38.tinypic.com/49nj2a.jpg">

Problem Advance QOS VOIP on ATM

hi,
I have a big problem for VOIP QOS on ATM interface.
Unable to have a proper voice on an ATM interface, no drop packets ...
I give you a link is to download a PDF that explains everything with schemas. Thank you in advance.

Pdf explains problem

VoIP and connection problems

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This page shows information about your ADSL connection if applicable.
Status
Configured Current
Line Status --- SHOWTIME
Link Type --- Fast Path
Operation Mode Automatic G992.5(ADSL2+)
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Stream Type Actual Data Rate
Upstream 1187 (Kbps.)
Downstream 11863 (Kbps.)
Defect/Failure Indication
Operation Data Upstream Downstream
Noise Margin 6.1 dB 6.5 dB
Line Attenuation 16.8 dB 31.0 dB
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Fast Path HEC Error STR 11420 80
Interleaved Path HEC Error NA NA
Error Seconds 2058 21
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Received Cells 26366448
Transmitted Cells 2181922

My ADSL2 broadband used to be 8Mbps down/1Mbps up (yours is 11Mbps down), and I didn't have too many problems with VoIP. I wonder if the backhaul from your exchange is overloaded (ie there is high contention)? I would expect quite stable pings from a site like the BBC. What is your speedtest like (try going to http://www.speedtest.net or http://speedtest.btwholesale.com/ from a PC preferably connected via a wire to your router rather than WiFi). Your line should be capable of roughly 10 to 11 Meg down and 1 Meg up. If you see much less than this then you may have contention problems. RegardsPeter PS: Your router shows "Fast path" but I know that the Brightbox 2 sometimes tells lies. Mine was showing that recently when it very definitely had been interleaved due to a line fault. This will add maybe 10ms to pings but should not affect transfer speed too much.

Voip Configuration

Urgent...i'm newbe on cisco networking, I have 3 2811 with voice router in my network, internet and data networking configuration now is good work, but i have problem with voip configuration, i can't do it. Scenarion for voip: all extension phone in one SITE can connected dial call to other SITE and all extension phone all SITE can dial call to out (ISDN). This is my append attachments my network topology. My router 2811 site B (Router B) have include call manager and interface to ISDN on SITE B and interface to PABX on SITE C is VWIC-1MFT-G703(E1 trunk).
Please...any one can help me to give example configurations or completion configurations for my network???
Thanks a lot for your Help....

Help me, you can do it friends????

E61 - Problem: Auto restart when Ad-Hoc Disconnect...

Hi,
I use E61. I connect internet with E61 by Ad-Hoc connection with my Laptop. When I disconnect the Ad-Hoc connection, My E61 auto restart. I don't know why. I tried many times.
E61
firmware: 2.0618.06.05
Thanks.

I usually connect to a router (infrastructure) but yesterday I connected to an ad hoc network and when I disconnected the E61 crashed and rebooted.
The WLAN features on the E61 need to be improved by Nokia as there are a number of problems:
1. crashes when disconnecting from ad hoc networks.
2. will only work with many routers when power saving is disabled on the phone and then the battery life is very short.
3. cannot connect to some secure wireless networks because EAP-TTLS-PAP is not supported.
4. the error message 'WPA Authentication Failure' appears randomly when you try to connect then a few seconds later it will connect without any change in settings.
The power saving would be a real problem with VoIP, but VoIP does not work with most providers!

VoIP Solutions

Hi All,
We are planning to have a VOIP solution in our organization with in coming months. We have 4 offices. Each office has E1 links between them. Our HQ has Cisco 3745 series router with 12.3 IOS running and other location has Cisco 3725 series router with 12.3 IOS.
I have some doubts, can any one help me.
1) Do I require any new hardware? If yes please specify.
2) Does my IOS support VOIP?
3) Is there any extra cost involved?
4) Can I use soft phone?
Regards

At all locations you probably have a PBX or Keysystem to support local office phones as well as the HQ site. Depending on the available interfaces on the PBX you would need hardware on the gateway so you can connect your 37xx routers to it. Normally connections are made through EM, FXS, FXO or T1 digital cards. These are VIC cards and you would need NM modules as well. For T1 you would need NM-HDV and VWIC T1 cards.
Your IOS does support the voip so you should not have any problem configuring voip.
Cost will be related to the hw that you would need to integrate your PBXs to the routers.
IP Softphone would need CiscoCallManager(CCM). Since you did not mention that you are expecting to install IP Phones at locations the above mentioned recommendations are tradition voip. If you are planning for IP Telephony support then you will have to look into:
CCM Server
CCM
IP Phones
And this will allow you to support softphone.
I hope that helps.
Javed

VOIP QOS over GRE ISSUE

Hi All,
I have this problem with VOIP phone calls been very poor, and need to fix it ASAP.
I have a stalite link between the Head office and Branch site.
Head office is cisco 3845 and Branch is cisco 3825.
And the Phone system is NEC ( with IP Trunk) via the GRE tunnel.
I have a GRE Tunnel between the sites since the IP address I am using on the LAN are Not supported by the sat Provider.
Can you please have a look on the attched and advice if I am missing any thing.
I have configured a QOS setting and applyied it to the tunnel interface per the following Cisco guide
http://www.cisco.com/en/US/tech/tk543/tk545/technologies_tech_note09186a008017405e.shtml
but the QOS polices are not Matching the NEC PBX traffic.
Thanks for your help
Nour

do u know what is the default marking of that traffic i mean the NEC?
what is the matching crateria u follow to match the NEC PBX traffic ?

How can you tell if the line has errors?

I'll try and keep this as brief as possible.
<background>
For 2years (up to 6months ago) I was on BE ADSL and was very impressed with the quality of the service (low latency, decent bandwidth) on my line. However when I heard that FTTC was being rolled out I switched to Zen internet on the understanding that I could then switch to the FTTC product as soon as it was available.
Immediately after switching to Zen I started to experience problems with VOIP (I use VOIP extensions against an externally hosted PBX) and was seeing CRC errors on the line. In the process (~2months) of getting this diagnosed the FTTC product became available and since I wanted uncapped switched to BT Infinity for Business Option 2.
</background>
I now find that again I get sporadic VOIP problems (but not as bad as on the ADSL2 service via Zen) but very regular BBC iPlayer dropouts listening to the radio.
Due to the previous problems whilst on ADSL2 I suspect I still have a minor line fault but as I can't access any stats on the line have no way to verify. Does anyone have any pointers on where to get line stats to see what is going on on the connection? I can't get any info out of the OR modem.
Regarding my network. I am using a cabled, gigabit connection directly to a Draytek 2820n which is in turn doing the PPPoE through the BT OR modem. I use two different machines and experience the same issues on both. I'm a server & network admin by trade so pretty sure it's not my internal infrastructure at fault.
My IP profile is apparently 38717 Kbps down / 10000 KBps up.
BT Speed tester shows achieved 26581 down and 4912 up though

jmansford,
I may be wrong about this as I am only refering to my own 2750n settings,but looking at your upload throughput test results some if not all of your problems could be due to a wrongly wieghted QoS setting and/or port capacity setting based on QoS.
As you line profile is ok at 38717 Kbps down / 10000 KBps up it looks like there is no problem on your actual line,but phone line noise test might find some crackle,but I don`t think so.
By your post,you say your are routing PPPoE through the Draytek into the OR modem.?
If so,maybe you could try leaving the OR or Draytek out of the chain as a process of testing where faults [if any] are.?
To sum up a bit imo leaving the OR modem out of the net chain might be a better way as I only use the 2750n on it`s own as the OR modem is not then needed.
Opps! sorry I might be wrong in my above thinking,but is your Draytek just a router or a modem/router combined like the 2750n.?
But either way the QoS setting are still important.

RV320 custom services management buggy

I've encountered several bugs and quirks, when trying to add custom services, and use them for firewall rules and port forwarding.
Including:
Deleting a custom service, does not (always) delete the one checked
Deleting a custom service, adding a custom service from port forward page, sometimes adds an "any service" rule
A deleted custom service, suddenly re-appears when adding port forward, even though verified deleted in Service Management
Duplicate services exist in combobox when adding port forward, though no duplicates are present under Service Management
I'm on firmware v1.1.0.09.
Are you aware of this Cisco? Is there a firmware fix scheduled?

Same Issue for me too.
I don't believe that Cisco had mess so much with these RV32x series....
Not only the "service Management" problems but i got other compatibility problems with VoIP providers.
It seem's that the NAT refresh time got some lake situation too, and unfortunately there is no option into the interface to change is value.
Give only few manual entries of service managment ports reduce this product into a residential usage, even this i meet these limitation with some residentials users with NAS that provide many services and absolutely need to open more than 30 ports.
Upnp can force to ad some entry, but in fact this can start another problem of overload the list in port forwarding
Cisco, can you read this, this is not an entry level of router we are talking here, and we can't effort to fall on basic services non work situation, please react fastest as you can to provide a thru fix on all these problems or we will need to completly forget Cisco Small Businness products routers.

Record time duration for fax in vxml

I am going to get the time duration used for fax sending in vxml:
<transfer name="mycall" destexpr="'fax://' + phone_num" bridge="false" connecttimeout="120s" maxtime="180s" cisco-longpound="true" cisco-mailtoaddress="'xxxx'" >
<catch event="telephone.disconnect.transfer">
<assign name="mydur" expr="mycall$.duration"/>
<goto next="#timeCount"/>
</catch>
</transfer>
<var name="startTime" expr="new Date();"/>
<var name="endTime"/>
<form id="timeCount">
<block>
<assign name="document.startTime" expr="startTime.getTime();"/>
<assign name="document.startTime" expr="Number(startTime);"/>
<assign name="document.endTime" expr="new Date();"/>
<assign name="document.endTime" expr="endTime.getTime();"/>
<assign name="document.endTime" expr="Number(endTime);"/>
<assign name="mydur" expr="document.endTime - document.startTime;"/>
</block>
</form>
But mycall$.duration is 0 after transferring fax.
And even if I use codes to count time duration, the duration, mydur, is about the same regardless of the number of pages sent.
Is it possible to use vxml to record time duration of fax?
Thank you!

Paste the following Config:
sh ver
sh run
debug of one call demonstrating the problem :
debug voip ivr all
debug vxml all

Spa2102 settings and router Netgear WPN824v2

I have problems with Voip Spa2102. When somebody calling me it showing that phone is busy but phone is not busy.
I thing i must change some settings in Linksys and my router Netgear. Can somebody help me with best settings?
Now i have: Internet -> Netgear router -> Linksys SPA2102
Firmware of Voip adapter if you need it is 3.3.6
And I living in Norway.

best way to isolate the problem is by connecting the SPA2102 directly to your modem...check if you are able to call out and receive calls...of course internet should be working first before you can use the phone to call out so connect a computer to the SPA2102. Once this is working just connect the Netgear behind the SPA..this will ensure that voice is always prioritized...
| isolate! isolate! isolate! |

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