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Just curius what yuo want to do with rtp ?
First solution has no any codec.
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I am trying to do a pbx for internal use... (voicemail, auto-attendant, conference bridge,...) I am currently using asterisk but will love to have the same in .net.
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Hi Ivar
Nice code.;)
I would ove to see the rtp implementation. I do have a project that truly needs RTP and could not find anything in c#.
Thanks.
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I like your job and i am so happy that there is a guy like you working so hard .
right now i am studying the code and i just want to use the SIP Stack .
i want to implement something on the SIP Stack , so i am wondering ,is probalby i can use the SIP folder to start working on , coz i found some many foloder like HTTP , FTP .cte and to be honest i dont need them right now , so i want you help in this coz i dont want to waste so much time to investage others code.
i hope to listen from you soon
Thanks
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Hi,
Some folders are related, so you can't skip all.
I dont see how other code disturbs you, you never need to look at these folders, if some relate code, you can later see it from SIP imported namespaces.
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Thanks for replay,
When you doing register for the SIP , you have to send the Invite Message to PBX switch , right , in my case i am using asterisk PBX switch , so i have to update their file sip.conf with the registaration information and the extension . then i have to make the call or invite , which part of your code doing this process , coz yesterday , i spent all the night study the SIp_core.cs and the regestartor method .
thanks
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Ok, currently i dont have register example completed, but i have: making a call and sending instant message example. Drop me email, i send it, i wont publish it before register is working too. NOTE: also current example wont allow incoming calls, you can call out only. If you ahev time, you get full SIP client, if all goes well, then RTP is there too with G711 codec.
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plz , i need that , so i have to impelement the full incoming sip calling by myself . cool .
making a call and sending instant message example can you send it to me .
sanaaliahmed@yahoo.com
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Ivar,
Congratulations on the great work! I believe that interest in .NET implementation of SIP will grow - it is just that the java world already has an open implementation and the amount of initial work to implement is great - your stack is very welcome!!
I've implemented an application using the stack to generate multiple simple SIP calls of varying duration and the stack is working well. I have discovered a couple of things that may be caused by my particular use:
I am calling dailog.Terminate() to close the SIP calls and under moderate load conditions, I will see a null reference exception at approximately line 416 of the SIP_ClientTransaction.cs file, which appears to be a race condition on the timer m_pTimerE, which is null. I have not investigated to determine the actual events leading up to the error, but put a simple null check around it as a short term fix.
m_pTimerE.Interval = Math.Min(m_pTimerE.Interval * 2, SIP_TimerConstants.T2);
I have only noticed this error on the E timer - all others seem ok. I am also noting many first chance "Object Disposed" exceptions (probably on the dialog properties) being reported, again probably due to my use of Terminate. these do not cause failure, but may be affecting performance and throughput.
I have also noticed that it is possible to put the stack into a "meltdown" condition (nearly 100 percent cpu) by issuing too many concurrent client calls - again, it may be my usage rather than a stack bug - I have not investigated this particular error any further yet.
I would be most happy to investigate further if that would be helpful.
Again, thanks very much for your work - it is very ambitious and quite complete and I look forward to more. I am also interested in any client code that you may have available.
Thanks again!
Don S
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Hi,
>I am calling dailog.Terminate() to close the SIP call. ...
Can you try latest ?
http://www.lumisoft.ee/lsWWW/Download/Downloads/Net/[^]
>I have only noticed this error on the E timer - all others seem ok. I am also noting many first >chance "Object Disposed" exceptions (probably on the dialog properties) being reported, again >probably due to my use of Terminate. these do not cause failure, but may be affecting >performance and throughput.
Terminate is right way to dispose, but after it most properties become unaccessible.
But there may be some bug in too, we can discusss it and you can send me exception stack trace, then its easy to fix.
>I have also noticed that it is possible to put the stack into a "meltdown" condition (nearly 100 >percent cpu) by issuing too many concurrent client calls - again, it may be my usage rather than >a stack bug - I have not investigated this particular error any further yet.
That stuff also interesting, some reproducer app would awsome, if yo have it ... .
Seems you are laso home in SIP, so drop me email, i give you msn ID, probably we have better to do live text chat...
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It seems to be biased towards making a proxy.
Is the interface for creating a client complete?
How would one go about this?
Thanks!
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Hi,
No it isn't. Its very general SIP stack.
Client application can make use of stack.
To prove that i'm currently working on SIP_Communicator class, what has basic instant messageing and call making methods, also i'm very close to complete RTP, so probably G711 sound calls can be made.
Are you interested in ? What you are tryig todo ?
But for unfortunately there is very low interest about .NET based SIP,RTP, i have got so few feedbacks ... .
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hum...... did i heard that you are very close to complete the RTP stack..... I will be very interested in that but i already told you keep the good work.....
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I would also be very interested in a .NET SIP+RTP stack.
After extensive searching, I too have been surprised by the lack of .NET solutions in this area... so you are somewhat a pioneer!
Please do keep up the good work - I would love to help with Beta testing it!
Regards,
Jason
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That's great! I know that this may be expecting a bit much at this point but will it be able to detect incoming DTMF (both in-band and out-of-band (RFC2833))?
I was hoping to try using it for an IVR application.
Thanks
Jason
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I think it runs on top of RTP audio codec(g711,g722.
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Hi Ivar,
I am also very interested in your .NET SIP,RTP solution. I am testing your demo as a stateless proxy and it seems working fine. I need to develop a VoIP audio+video network within LAN (within one house) that would connect both HW and SW VoIP clients. I could probably use your SIP_Communicator class as a basis of my SW VoIP client. Has it been already completed?
Regards Marek
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Hi,
There has been lot of changes in SIP,RTP, lot of rewrites.
Currently i'm rewriting RTP, 90% done, code finished very soon, but testing takes wee time probably.
SIP is pending a little, new classes implemented for creating phone app(SIP_UA).
Call establishment,termination,IM works, but lack of RTP completion, reINVITE and SDP processing not implemented yet.
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I'll try do my best, but user must help too, like reporting bugs, giving feature requests,ideas, ... .
I'd like to add presence server too, then most of features there.
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Hi Ivar,
First, congratulation on this one!!!
I tested it in different way and one does not work:
I have a server with a public IP (No NAT) and 2 clients behind 2 different NAT (different location) -> they can call each other but no sound.... it seems that the RTP does not get the translated address but the SIP does....
The others test worked fine (like 2 clients in the same NAT).
Let me know what you think....
Yann
PS: You still got my 5!!!
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Hi,
Current SIP proxy never modifeis RTP, so what comes in mind:
1) If proxy has natted public IP, you need to fill host name or at leat put public IP there.
2) You havent specified STUN server in SIP client.
What sip client you used, cureentyl i use X-lite, hardware Siemens SL75 WLAN, Linksys SPA 3102 pstn gateway.
Please be free to ask more, i'm willing to debug out whats problem.
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Thanks for your answer!
The proxy server as no NAT, it has a regular public IP (207.179.xxx.xxx).
As client, I have tried with 2 x-lite (I did not specified the STUN server because they autodetect it.)
I tried with diferent IP phones too (Linksys, Sippura, Cisco, Grandstream)... they all works with Asterisk, SER, LCS but not with your Proxy.
PS: if you want me to run some interoperatibility test for you for the future, I have plenty of SIP Hardware, Servers with Public IPs, and SIP <-> PSTN Gateway to test with.
Yann
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X- won't autodedect STUN, you need to specify it. Like stun.counterpath.net or stunserver.org.
Also if you set Host NAme in proxy to domain name, or if no domain name, use public IP, this
ensures ACK command reaches UA1 to UA2. Probably if you do both(specify stun and host name) all works as you expect.
Same goes for phones.
Why asterisk works, probably you run it B2BUA mode and RTP goes through it.
What NAT router do you have ? SO far i haven't meet any what won't work.
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