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Hi
I am new to this form.
I had a look at the tsril.zip example as it seems from this forum that to make/receive a data call the ril is the only option. Unfortunately the documentation for ril is nonexistent and I want to ask if someone has done an example of a client/server data call, data transfer between to XDA's.
Thanks in advance.
Anton
It all depends on what type of data you want to transfer. I have never used RIL so I don't know the advantages of it, over the normal programming interface. I say normal, not thinking there is anything wrong with RIL but the little I do know is that RIL is operating system level and is normally hidden from the application programmer. There must be times when RIL must be used if Microsoft forgot functions in the normal APIs.
However I would say that for a data link between 2 devices then RAS seems the way forward. One device would be the Server and the other the Client. I have only ever programmed RAS from the client side, making a Telnet link to an already running server.
I just had a look at the help on Embedded Visual C++ 3.0 and under 'Creating a TCP Stream Socket Application' it explains the server side and then links to the client side. Have a read there.
RAS is not an option as it is a server consept, which I think does not even exist as a service on the desktop windows os let alone on a Pocket PC.
What I am looking for is something similar that you would be able to do with a landline modem or a gsm modem connected to a serial port of a pc (even a usb modem) which is to send AT commands to make/accept a call and then to make use of the serial port as if it is an io stream. This is possible on windows, linux using either c#, c++ or java. I can even do this with an embedded gsm module like the siemens TC45 java module.
What I can not understand is why Microsoft and the XDA suppliers (I am using a Qtex running Pocket PC 2002) is making it so difficult to make/answer a data call and let you run your own protocol over the connected stream.
RAS (Remote Access Services) is built into all Windows Operating Systems, including Windows CE. Did you look up the help example I pointed to ? What you do with RAS on an application level is for you to code but the connection itself is handled by the operating system.
As to making a connection similer to a modem and using AT commands. Then no you can't use AT commands directly, but you don't need to. To get that type of connection you need to use TAPI. Once Tapi has made a DATA_MODEM type connection, you use the TAPI callback connect event to then ask for a file handle that you can use with the normal WriteFile and ReadFile commands.
I already have something on this forum about that see:-
http://forum.xda-developers.com/viewtopic.php?p=7857&highlight=#7857
The problem is not so much making the data call but it is accepting/answering the data call that I can not get working on the XDA
Not sure if this is your problem, but in Australia they have a seperate "data" phone number for the same SIM. If yo call one number you get voice, if you call another number you get data on the same sim. Not sure if this may be effecting you.
I just looked into all this ras stuff a bit deeper, and yes I think I may of been talking rubbish. Although RAS does exist in PPC2002, it can't see any functions that allow the device to answer the line. I can see that PPC2003 has a Ras Server and RasIOControl that looks like the answer there.
Also looking into Tapi, I can't see how you answer a data call using that.
So I now understand your problem. It seems PPC is made as a client device.
Is GPRS the answer ? With that the link is always present and then you can use the built in Ras functions on seperate devices. Since you only pay for data actually sent then would it matter if the network link was always present ?
I have only worked out Tapi and Ras from the client side so I have never had this problem, but I agree it is an interesting one. I will have a bow out of this thread for a bit and see if anybody else knows how to Answer a Call.
Thanks for trying.
GPRS is also not the answer as you need a server in between that both XDA's can connect too and use as a router as the XDA's do not get fixed IP adresses but actually a NAT adres from the APN so you can not connect directly between the two.
Now that's an interesting thought, how about using some free web space to act as a pigion hole for your data ?
Believer: A seperate number? How do you know this, and what would the number be?
I can actually make two XDA connecting to each other using TCP. The trick is that I have one client always connect to a server to register it IP address with an ID. Then the caller send a query to the server to look up the callee's IP address.
In this way, accept() and connect() work fine.
I am using AT&T network and not sure about if other networks behave the same.
I wrote two application using TAPI. One is ModemDial and the other is ModemWatch, if I make a voice call the ModemDial dials the number successfully and ModemWatch reports that an incoming call exist. But when I change the behavior of call to DATAMODEM, the ModemWatch couldn't track any incoming call.
Is there any one, had some experience with pick a call for DataModem?
Best regards,
A. Riazi
riazi said:
I wrote two application using TAPI. One is ModemDial and the other is ModemWatch, if I make a voice call the ModemDial dials the number successfully and ModemWatch reports that an incoming call exist. But when I change the behavior of call to DATAMODEM, the ModemWatch couldn't track any incoming call.
Is there any one, had some experience with pick a call for DataModem?
Best regards,
A. Riazi
Click to expand...
Click to collapse
In your ModemWatch application make sure you are calling lineOpen with dwPrivileges (the 7th parameter, 1 based) equal to LINECALLPRIVILEGE_OWNER and dwMediaModes (the 8th parameter) equal to LINEMEDIAMODE_DATAMODEM.
If you're able to answer a voice call successfully, making these changes to the ModemWatch application should allow you to answer a data call.
Hi,
Anyone knows any free VOIP softphone for O2 Exec which allows to configure to use third-party SIP providers (e.g. sipdiscount, sipgate etc.) ?
Thanks!
SV
use skype
Sorry, I meant to configure with SIP protocol.
sjphone, works fine
With SJPhone, at the moment, when calling, you have to use loud speakers or headset not the phone speaker. Is it right?
Thanks!
That happens with any phone edition device.. Apps don't have access to the other speakers...
Re Sip
I have tried many. However most free products are generic to a provider,
Successful products are:
Xten lite from xten Networks
SJ Phone this is the best and is easily configured to multiple profiles.
It intergrates nicely with our office Sip server too,
Regards
DW
Re Sip
I have tried many. However most free products are generic to a provider,
Successful products are:
Xten lite from xten Networks
SJ Phone this is the best and is easily configured to multiple profiles.
It intergrates nicely with our office Sip server too,
Regards
DW
svkhtn said:
Sorry, I meant to configure with SIP protocol.
Click to expand...
Click to collapse
May I know what is SIP protocol?
Is it different with Skype protocol? Is SIP protocol is a standard one?
Thx.
Regards,
Arto.
Artosoft said:
May I know what is SIP protocol?
Is it different with Skype protocol? Is SIP protocol is a standard one?
Click to expand...
Click to collapse
SIP is a standard - most VOIP application and hardware uses it.
I meen: SIP RFC 3261
Skype is a own protocol and only skype clients can connect each other using skype soft phone or by skype gateway.
I guess that the source is not public.
I use AGEphone.
http://www.ageet.com/us/
It is not Free.
But, maybe best.
Files are light, but works very well.
the sound is very clear.
We can use SIPdiscount, VoipBuster, VoipStunt, StanaPhone, SIPphone, Free World DialUp, Agilephone, etc all at the same time.
I installed SJPhone and could get connected. I could dial my cell phone number and it will ring my cell phone.
But there is no sound whatsoever in my Universal/JJ.
Can anybody help me? Did I miss something? I tried to plug in my headset but no luck as well.
Thanks.
Also, may I know which version of SJPhone you guys use?
Can you send the link for the download? I can only find the one for 2003SE and 2003, nothing for WM5.
Since I didn't see a follow-up post by anyone, I'd like to say that the SJPhone for Windows 2003 SE works with the universal *only in portrait mode*.
Has anyone [other than shamilsh] tried the ageet soft phone suggested in this thread?
Does Skype work over GPRS or just 3G?
Regards.
Just for text messages, skype is ok with GRPS - but for voice you need 3G oder WLAN.
Hi
Any sip cliente for wm5 that have G.729 (8 kbps) or G.723 (5.3 & 6.3 kbps codecs ?
sj phone only 711 or their own gsm....
Owl said:
Just for text messages, skype is ok with GRPS - but for voice you need 3G oder WLAN.
Click to expand...
Click to collapse
Is that need 3G as in would be slow, or need as in can't do it...
Just that GPRS is free for the T-Mobile plan I'm using, but think I have to pay for 3G (Relax + Web & Walk thing)
are there any programs that allow routing of the sound to phone speakers instead of main speakers?
Guys....just use sj phone with sip service from voipbuster.com and forget the rest.... and now you are talking....FREE and with free local rate call back number also...now...that ROCKS :twisted:
BR
Hi guys
I know many people have had one way voice issues using the various cabs for enabling the WM6 VoiP client but I don't think this issue is related to that.
I'm connected to a SIP PBX over an OpenVPN connection and everything connects but I the other party cannot hear me. I've done a packet capture and although the SIP INVITE is coming from the correct source address if you drill down into the SIP packet the owner creator etc is the original IP of the device (not the VPN one).
As a result the RTP stream is being directed towards the wrong IP and I'm getting one way voice. This issue doesn't happen with 3rd party SIP clients but I haven't found a good one yet (fring doesn't work on our PBX).
Has anyone came across this issue before?
After further investigation the WM6 client works perfectly when used over a wireless access point. The problem only occurs when using OpenVPN as the original IP address of the wireless adaptor is sent in the invite packet (rather than the VPN IP address), this causes the audio to be streamed to the wrong IP and as such one way voice is experienced. If I find a workaround I'll post it up.
shippyt said:
Hi guys
I know many people have had one way voice issues using the various cabs for enabling the WM6 VoiP client but I don't think this issue is related to that.
I'm connected to a SIP PBX over an OpenVPN connection and everything connects but I the other party cannot hear me. I've done a packet capture and although the SIP INVITE is coming from the correct source address if you drill down into the SIP packet the owner creator etc is the original IP of the device (not the VPN one).
As a result the RTP stream is being directed towards the wrong IP and I'm getting one way voice. This issue doesn't happen with 3rd party SIP clients but I haven't found a good one yet (fring doesn't work on our PBX).
Has anyone came across this issue before?
Click to expand...
Click to collapse
This is a known SIP issue. As SIP had been designed to work over end-to-end connections. As SIP is a self routing protocol, the SIP server and user agents use the source IP stated in the SIP header instead of the source IP stated in the IP header for their routing
Most 3rd party SIP clients use a technique called STUN to discover their global IP (behind the NAT, or as in your case the VPN local IP) and they put that IP in the source IP in SIP header.
Tokko said:
This is a known SIP issue. As SIP had been designed to work over end-to-end connections. As SIP is a self routing protocol, the SIP server and user agents use the source IP stated in the SIP header instead of the source IP stated in the IP header for their routing
Most 3rd party SIP clients use a technique called STUN to discover their global IP (behind the NAT, or as in your case the VPN local IP) and they put that IP in the source IP in SIP header.
Click to expand...
Click to collapse
is there any workaround? I do not want to install STUN server on a local pc. I connect to my router openvpn (dd-wrt firmware). mega version has also a milkfish sip server. I have been thinking maybe I could use it as proxy to resolve this one-way voice problem?
Hey, I had the same issue recently and I have found a solution: do not enable VoIP calls over 3G/GSM when using SIP Config Tool. I was actually able to make and receive VoIP calls over an OpenVPN connection - the sound was a little choppy at the callee's side, but in general it was quite good. The only problem I have so far is that the phone unregisters itself from server when going to standby mode and registers back when turning on, so I can receive calls only in the latter case.
shurik_1 said:
is there any workaround? I do not want to install STUN server on a local pc. I connect to my router openvpn (dd-wrt firmware). mega version has also a milkfish sip server. I have been thinking maybe I could use it as proxy to resolve this one-way voice problem?
Click to expand...
Click to collapse
Not to mention native SIP client you are using probably doesn't support STUN anyway.
Proxy idea sounds doable, but then WM device is still going to use the ip of its wireless interface in the INVITE message. Unless this is fixed by the router, milkfish will still be unable to encapsulate incoming RTP traffic into the VPN.
Perhaps you can mangle SIP INVITEs with sip_conntrack available at http://www.iptel.org/sipalg/?
What application do you people use here for making true voip calls (not skype)?
On my old XDA 2S I used to have something that was nicely intergrated in the system with the phone interface, and that I used with my Asterisk setup.
Unfortunately I can not remember the name of the application.
How about the Voipbuster Mobile???? http://www.voipbuster.com/en/mobile.html
Nop.. I want to use it with my own server.
The Voipbuster client will only work with VB.
Hello everyone,
Is it possible, on an Android phone, to forward incoming calls (from the GSM network) to a local asterisk server (via SIP)?
I know that Android supports SIP as an extension, but that is not useful to me. Currently, the only way I've found, is to use a Bluetooth adaptor on the asterisk server. I'd prefer the SIP solution instead, if one existed.
Thank you!
Andr0ides said:
Hello everyone,
Is it possible, on an Android phone, to forward incoming calls (from the GSM network) to a local asterisk server (via SIP)?
I know that Android supports SIP as an extension, but that is not useful to me. Currently, the only way I've found, is to use a Bluetooth adaptor on the asterisk server. I'd prefer the SIP solution instead, if one existed.
Thank you!
Click to expand...
Click to collapse
I've got the same question.
"Y U dig up old thread?"
See above
So far, I've only found information, that you COULD compile Asterisk for Android. And that the RIL WOULD be difficult to handle.
Has anyone ever tried it? Did you manage to get it working or not?
No I couldn't find a way to do it. In theory, someone could create an app that redirects a call to the PBX, but nobody has done it, afaik.
What I've done instead, I've got a Grandstream VoIP phone with bluetooth support. So all my incoming mobile calls actually ring on the Grandstream. While the calls don't go through the PBX, at least I don't have to bother with the mobile phone anymore.