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Hi
I have downloaded Simple SIP and i am not shore how to set it up, i have 2 PC, one on 192.168.0.3 Local
and one remote 192.168.1.8
I have tried to set From sip:192.168.0.3 To sip:192.168.1.8 and on the remote side From sip:192.168.1.8 to sip:192.168.0.3 but i can´t get them to answer any call.
Am i missing something ?
>For WavePlayer and rtp audio, first you need to sort out if mic or play out causing delay.
>There are no adaptive sound buffering, so i can't see how delay increases ...
Well it works perfectly without any delay when i use IPSOUND with the same input / output so i don´t know.
Regards
Pete
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You need to use "sip:user@ip:5060" as To:.
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Hi Ivar
Strange, I cannot get the them to answer I have tried this settings.
Local machine
-------------
From sip:blue@192.168.0.3:5060
To sip:black@192.168.1.6:5060
Remote Machine
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From sip:black@192.168.1.6:5060
To sip:blue@192.168.0.3:5060
No one of them answer any call
I haven't look so mush at the code but are they listening to the From: IP when they start ?
I have tested to connect from both PC to echo@iptel,org and music@iptel.org and that is working
but they do not connect to each other as above.
//Pete
modified 23-Jan-12 16:50pm.
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Hi,
Do you get phone ringing ?
In one(which) computer you may leave all to from as is, on another you only need to alter to field:
sip:xxxx@remoteIP:5060
Phone listens on port 5060.
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Hi
Yes it ringing, text message says calling and i hear a tone every time, but there is no response on the other side.
Strange this, i have also tested on same network lan, ie 192.168.0.6 and 192.168.0.3, (not over VPN)
but there is no connection there either.
I have copied the files that are in the Bin/Debug to the other computer and the program run ok, on the other side i run it from visual studio, but i don´t think it can have something to do with that.
I will get up wireshark to se if i can se any IP traffic.
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I suspect the problems begin when multpile active network connections.
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I have look at wireshark and it seems to some network problem, it says Destination unreachable.
the packets are as follow.
Quote: 177 32.487319 192.168.0.3 192.168.0.6 SIP 429 Request: OPTIONS sip:ping@publicIP.com
Quote: 178 32.488903 192.168.0.6 192.168.0.3 ICMP 190 Destination unreachable (Port unreachable)
Quote: 180 32.897771 192.168.0.3 192.168.0.6 SIP/SDP 702 Request: INVITE sip:bl@192.168.0.6:5060, with session description
Quote: 181 32.899273 192.168.0.6 192.168.0.3 ICMP 190 Destination unreachable (Port unreachable)
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Hi,
I've been playing with your code and need to build a TCPServer to complete a demonstration I am making.
While browsing on your website (http://www.lumisoft.ee) I saw that you have a new version of LumiSoft.Net library.
I tried to use the compiled DLL with this project and it seems that you've moved things around and might have placed them as obsolete.
Is there a new version of your AdvancedWave sample with your new code?
Thanks,
Noam
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Thanks Ivar for the fast reply.
I played with it a little bit and got a few exceptions...
To do that simply start the server with your IP's (I used the local IP and 127.0.0.1) and start sending the mic data.
The sending microphone dialog box appears and then pressing start, waiting for a while then stop, then start again, waiting and stop will result with a few exceptions (or at least it was for me).
The error I got was :
Unhandled error: System.InvalidOperationException: Handle is not initialized.
at system.runtime.interopservices.GCHandle.AddOfPinnedObject()
at LumiSoft.Net.Media.AudioIn.WaveIn.<>c_DisplayClass1.<onwaveinproc>B__0(Object state) in E:\LumiSoft\Net\Net\Media\AudioIn.cs: line 693
at System.Threading._ThreadPoolWaitCallback.WaitCallback_Context(Object state)
at ....
Any suggestions?
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I dont know if helps, but i updated rtp demo to latest.
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If you did it today after 14:15 (UTC+2) then I should get the new version.
Otherwise....
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Hi,
I was just wondering if it was possible to achieve this functionality (of the encoding and decding) uisng NAAudio
http://naudio.codeplex.com
Thanks
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Hi,
Pobably thats possible, if you are talking about u-law,a-law.
That codec inputs 8Khz 16bit raw PCM audi and converts to 8khz 8bit.
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Hi,
Im looking for equivalent methods for
G711.Encode_aLaw
G711.Decode_uLaw
etc
in the NAudio library but not sure how to do it?
ANy ideas?
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You nee by your own or use some lib.
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Is it possible to listen to audio received from multiple sources...
Like have communicate with two clients using your code?
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You need to listen each audio source on it's own TCP port, then it's possible.
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tnx for your code & sample
how do i broadcast audio strem to many client.
i want multicast strem on the web client
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Hi,
Muliticast not possible for public networks, multicast only works on local area network.
So sending data to multiple recipients, you need send multiple data streams(at least 1 per recipient).
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tnx for you.
how do i use this sample for all client in my local network?
can i send audio to all client in Lan?
if i can use it, what's ip address for local & target?
tnx
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Multicast address range is: 224.0.0.0-239.255.255.255.
You can use any of the address in range, but all clients must use same address !!!
Also you can have 1 sender and multiple listeneres, you can't have multiple senders.
To support multiple senders at time, you need to use RTP. Also you need to do audio streams mixing.
Also you need to set socket option for example:
socket.SetSocketOption(SocketOptionLevel.IP,SocketOptionName.AddMembership, new MulticastOption(IPAddress.Parse("224.0.0.1"));
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Great application library.
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Hi Ivar,
Would firstly like to thank you as this library is great and very useful. I am developing an application where there is one server and multiple clients who all listen to what the server has to say. I have got it working so far but have noticed that, through UDP, sometimes packets are dropped or buffered and then they are played back after the connection is re-established. Which is causing a delay between multiple clients. I was wondering if there was a way to disregard any packets that are delayed and to continue playing whatever is coming out at that time. This way even if one client missed some it would not care and would still be playing the same audio data as the rest of the clients.
Kind regards,
Stuart.
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