Having spent the last 30 minutes having a search around this topic does anyone have any info on lack of audio on the Hermes for a SIP VoIP call? Is an upgrade needed?
I get feedback when I place the phones back to back so the audio is there somewhere but it doesn't seem to pass speech through the speaker of the hermes.
I am using a Movial Soft Client and an IMS Server for the SIP proxy functions and have no problems PC to PC (whether NAT or not).
If this has been answered before sorry I couldn't see the wood for the trees!!
most probably it's a sip problem when you are behind a NAT. Configure a STUN server on the SIP client and should be ok.
Related
My first post in this outstanding forum
I've spent quite some time looking for programs able to run my SIP VoIP desktop client on Tynt. On WinXP I use X-Lite with VoIPDiscount and I know for sure it's possible to configure X-Lite with many other SIP based services like VoIP Stunt.
Is there a tested client for Tynt? All I've found was for PPC2003 and I got errors all over. Ideal'd be a client working with both UMTS and WiFi, but WiFi would suffice completely.
On the other hand, Skype works flowlessly
I use SJPhone:
http://www.sjphone.org/preview/ce/
You need to adjust settings to make it work correctly, also audio is a lot better with the USB headset, see this thread for right settings:
http://forum.xda-developers.com/viewtopic.php?t=59131
You can also use x-lite CE 1.01, find it here:
http://forum.xda-developers.com/viewtopic.php?t=58154
SJPhone all the way here too. It also works as a Vonage Softphone!
thanks guys. I saw SJphone but somehow I didn't manage access to the beta page.
I'll let you know
If your provider supports IAX protocol (which is much better while on hostile firewalled networks) my choice is http://www.voipsurfer.net/
My sincere thanks, guys I'm gonna try them all. At this point I created correctly the profile with SJPhone; the same profile works great with the windows client (after creating the profile I copied it to my desktop and the XP client accepted the profile like it was its own). After applying the profile the PDA client asks for id & pw, but I cannot register. Explorer is online so I know the internet connection is fine. I think I need some working yet.
I have managed to instal the pocketpc version of xlite and it works ok from the point of view of me trying to ring my home phone, it connecting and disconnecting ok and it registering as a broadband phone call through my isp when I check my online records.
The sound quality on the device is unusable.
When I use the supplied mini usb headphones I can hear the person I am calling very clearly but they can not hear me properly.
Can someone talk me through anything I can do to improve the settings so that I have a chance of using my device for voip calls whilst at home. I am not in the realms of anything to techincal though.
Hello everyone!
Situation: I have several VoIP provider (sipgate, 1und1...) My problem is I can only setup a single provider in my phone with built-in WM6 VoIP client.
Solution attempt: I set up an account with pbxes.org. There I added my SIP providers and an extension with username and password. Now I tried to set this up using the Sip Config Tool V2.0.1. Here are the settings:
Sip Server: pbxes.org Port: 5060
(tried others like sip.pbxes.org(com) and the IP)
username: username-extension_number
password: password
That's it. It's not working. The Today Internet-Calling-plugin tells me "searching....", but never to "selected"
I also tried using x-lite on my desktop pc. With the same settings I successfully registered and I was able to use pbxes just as it is supposed to be.
When using sipgate directly with WM6 it works, but not with pbxes.
I am kind of running out of ideas. So any help is really appreciated.
Thanks in advance,
Stefan
Tried several times as well on my S620.
Now, two days ago got a S710 and was curious to test VOIP on it.
Very first time I tried I was thinking everything was fine, at least nearly.
The Today plugin showed "Searching" (like with providers it works with)
The icon for Internet Phone showed up.
Unfortunately it "semi worked" only that time, any other tests I tried the Today Plugin didn't even get into "Searching".
The bad thing is that it did work once, like meaning it could work. Now PBXES status shows as Registered but it isn't actually. The other bad thing is that I tried to post in their forum for help but didn't receive much info back.
Bad luck, but I'll try and try again. Maybe newer files one of these days and our probs would be solved.
Ka.
Subscribe message
I ever check the message from sip server, I found that windows mobile voip client not only send register message, but also try to send subscribe message, I guess that it is reason that some sip server only may work as registra, and not handle correctly subscribe. it seems the reason the client always display "searching", I will try to modify sip server to support subscribe message in the future.
try fring
try using fring...
it supports multiple sip accounts.. and actually register u not just subscribe you.
You are right, Fring or any other 3rd party app connecting to a SIP server can work maybe better than the "builtin" SIP functionality but personally would prefer being able to use a solution where I dial like usual and calls get automatically router to VOIP, not having to switch to a separate app.
This is why I'd rather like the builtin sip stack to work on its own with any SIP provider. I tested it with PBXES but doesn't work. I'm currently renting a VPS with Asterisk (Trixbox) installed and it does connect. Only prob is that even if it manages to register, seems MS VOIP implementation is still not really compatible or suitable for the use I wish to have.
Ka.
This is an old thread but this is the same problem I have on my HTC HD2 (Leo).
Installed sip drivers and voip config cabs and configured it with my account through pbxes and it is not working.
Can someone help?
Android (Moto droid, HTC Hero) I had no problem with pbxes and sipdroid...
Hello world.
Since I can't hear the other party while placing VoIP calls through the integrated SIP client of WM6 via FON hotspots (probably due to the double NAT-routing on most hotspots), I'd like to know if there's any possibility to implement STUN support.
regards,
Inquisitor
Hi there, I think you should post your question in this blog:
http://blogs.msdn.com/cenet/Default.aspx?p=3
I'm bypassing the issue by using OpenVPN on my S710 which connects to my home gateway.
STUN Support in 6.1?
I have been searching around quite a bit.
Is there any support for a STUN server with the VOIP "Internet Calling" Feature on some of the newer ROMs. I'm on a Kaiser and it seems to register. I just can't seem to make a call to an asterisk system.
I believe this is because I am behind a NAT, so I think STUN is required for it to work properly.
doing same but no voice
jockyw2001 said:
Hi there, I think you should post your question in this blog:
http://blogs.msdn.com/cenet/Default.aspx?p=3
I'm bypassing the issue by using OpenVPN on my S710 which connects to my home gateway.
Click to expand...
Click to collapse
i m also having the same issue of no voice, could you please guide on how you accomplish this, my current setup is like this,
i have my server configure as openvpn server that is behind a router, ssh, ovpn ports are forwarded to server, ovpn client is xp and which is behind another router, xlite as softphone and working perfectly, ovpn installed on pocket, vpn tunnled is connect can connect through to server using putty, pocket pc is showing signal means registered with asterisk server (openvpn server is same) can dial also and it shows line connected as well, but no dial tone and no voice.
linux server with asterisk and openvpn
client htc universal (jasjar)
i hope i've given a clear picture of my setup, thanx in advance for any help...
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/?
Hi,
From Internate search I found this forum which is having very useful research and Discussion. I have one question and need solution and reply setp by step for the following issue:-
That I am trying to send voip call by using IP 192.168.0.20 using Port:6800 as SIP port to the Gateway from VPS (softswitch). At the destination we are using a Router (Dlink) for connectivity of internate and Gateway.
When I configure GW with IP 192.168.0.20 SIP port 6800 with username and password, and tried to connect Gateway as SIP client from Softswitch, the messages comes "The GW/Registrar Client does not register as SIP client". Thats means switch is not reaching to Gateway.
As I understand from some study that SIP-based communications cannot reach LAN users behind firewalls and NATs automatically because firewalls are designed to prevent inbound unknown communications.
I would much appreciate if any colleague from this forum may guide me step by step and let me know how this scenario can works.
Thanks
Sameel
I've Googled it and there are two ranges of ports you need open for it to work. Usually though ICE/Stun can get past that. Have you tried that?