Hi all,
i developed a VoIP application in Symbian C++ my application don't use Defaul VoIP Profile , i want to know if is a possibility for my application to intercept a call and if application is...

hi ! i have same problem with g729 coding, i can coding and decoding frames , i deleted header ( first 2 bytes from APS frame) and when i want to play it on phone i hear a noise when no data...

In API is not specified nothing about this header , i wold cut it, copy to buffer and then append back to incoming data, but he is changing in streaming process , i see APS is for interconnecting 2...

Hi all
I have a problem with streaming audio from APS, if i streaming in two APS servers ( 2 different phones) i can play normaly but whit others i hear very big noise, i change all formats and the...

Hi , i have a problem with RTP stack , i develop a VoIP aplication add my rtp don't work very good,
i use APS and RTP Api from plugin.
I get error USER 23 when my codec and remote codec is the...

Hi , i have the same problem , i trying all expressions bunt i have response 480 Not Available

my capabiltes is:


_LIT8(KCapabilities,
"<SIP_CLIENT ALLOW_STARTING=\"YES\">\
<SIP_HEADERS>\...

i see my question is unique ....

i create new Number Editor, for my problems but now i have 2 others
Firstly i don't know how deactivate this , i can change focus but i cannot delete it from...

Hi all!
I have few question with number editor
i create a number editor dialog and when i Run it i have some problems :

This is dialog:

CAknNumberQueryDialog* Dialog =...

Hi all!
I have a big problem , and i don't know how resolve it.
In my VoIP application on SIP, i don't catch any request, on Emulator when i send invite to it, he send 100 Trying , after that 404...

you first must use observer class MProgressDialogCallback
class CYouClass : public MProgressDialogCallback
{
....
}

and use function :
void CYouClass::DialogDismissedL(TInt aButtonId)
{...

my question is , who send 404 Not Found ? may be in Emulator i don't have port binding with my application, but why my phone crash when i call him ?

i write a SIP VoIP application without profile by Nokia, and my registration bindings works fine, but when someone call me, i don't catch INVITE and any Incoming Request. in wireshark i see incoming...

And is interesting ... if i create INVITE request and after that i delete iConnection, when re-register , conn.GetLocalAddrL(iLocalAddr) give me a panic, if i don't create any requests only register...

Hi i have a question with my unregister function !
I must unregister my sip application, i don't use Profile APi, and i delete all object , in order;

delete iClientTx
iClientTx = NULL
...

problem is resolved ..
thanks

I resolve problem with my application and my server, I can register , and invite .
tnx to all

i have the same problem !
i posted my invite message http://discussion.forum.nokia.com/forum/showthread.php?t=136279

can someone help ?

Tnx , problem was in uri

Now i have other problem , when i send INVITE , server send 100 Tryng , then 403 Forbbiden , i don't know what is wrong in my paket.

i make call from 349 to 339 ,...

Hi!
I writhed a small SIP Client without profile by nokia , but when i select register function i have this error:
KErrSipCodecPort -17767

In this forum i don't find any post with SIP Codec...

I find problem , is in 'conn' object , if i change the ConnectionL() function and i create connection in Register() function, RegistrationBinding work but then connection is not active :( , what i...

Hi !
I developing an VoIP application on s60 fp1 , and i have Error when i send REGISTER mesage,
KErrArgument -6 An argument is out of range or Bad request
i don't know what is the problem, pls...

Tnx, finely works ! i change the network connection for both endpoints and works. intresting is that when the endpoints have two different networks i don't hear anything but he hear me very good....

i find this exemple is very simply and is that you search
http://wiki.forum.nokia.com/index.php/Image:RTPexample.zip

Hi
I have problem with Nokia sip phone, any codec don't work , and i don't hear anything, i change the order of the codecs and in Wireshark i see normal connection.
my device : nokia n95

I can register and send INVITE on nokia n95 , but when connection is enabled, i don't hear anything but who INVITE me or who i invited hear me very good.
i instal
APS 2.4
voip setings and the...