FAQ

Q. How can I create a multiline softphone? I don’t see the support for multiple SIP accounts now?

A. Please use a separate (T)AstaSIPEndPoint instances for each SIP account. Please give them different BindingPort’s values.

Q. The outgoing audio from my softphone is good, but there is no incoming audio.

A. Please try to use

Q. I want to pass custom headers to call. How can I do it?

A. Please see MakeCallEx method. There is parameter named “Headers”. It must include header as shown below:

HeaderName_1: HeaderValue1

HeaderName_2: HeaderValue2

…

HeaderName_Last: HeaderValue_Last

The lines are delimited with /r/n codes – or 13 and 10 in decimal (#13#10 for Delphi).

Q. How can I get the incoming call headers?

A. Please see GetCallInfo method. The last parameter is out parameter – headers are returned as lines delimited by /r/n(13 and 10 decimal).

Q. Does ASTA SIP support both DTMF in band and out of band detection e generation.

A. Yes

 
Q. I’m trying to use IM/presence methods. My application subscribes to presence info using the RemoteOpenPresence method but doesn’t get any callbacks. I looked in log file – SIP 200 OK response is here.

A. The OnRemotePresenceOpen will be fired on the first presence info from remote side. So if you are developing the remote side application too – please call UpdatePresence after AcceptPresenceSubscription.

Codecs FAQ

Q. What audio codecs are supported by AstaSIP SDK?

A. A. G711a/u, GSM, G729, G723.1, iLBC.

Please note – you are responsible for licensing G729 and G723.1.

Q.  How can I use my own codec?

A.  You can use your own external RTP/media library together with AstaSIP SDK or contact us – we can add your codec as an external DLL.

Q.  Codec management methods requires the codec’s name as parameter. What are valid names?

A. G711-u  codec is named as PCMU, G711-a – PCMA, GSM as GSM, G729 as G729 and G723.1 as G723.1

Q. What is the initial order of codecs?

A. G711-u, G711-a, GSM, G729, G723.1. We recommend to set your own codec priorities on program start.

Q. How can I know what codec is used for conversation?

A. Please see OnCodecChosen event description

Q. What is expected order of events during call transfer?

A. If your softphone calls TransferCall the following events will fire:

- OnTransferAccept - it means the remote softphone received a transfer request and is reviewing it. It does not mean the actual transfer finish - it does not mean the transfer even starts however

.

- OnTransferRejected - it means the remote softphone does not try to make a transfer  .

Well, if your application gets OnTransferAccept then the existing call may be terminated as all responsibility is on the remote softphone.

However there is a way how your softphone may get information about transfer results. Please check these events:

- OnTransferProgress - it means the remote softphone calls to its new callee and gets provisional/early media packets from it.

- OnTransferSuccess - it means the remote softphone made a new call ok.

- OnTransferFailed - it means the remote softphone cannot make a new call.

These three events will occur only if the existing call is not terminated in the OnTransferAccept event's handler.

Q. How can I modify the SIP packets on the fly?

A. You should to call the SetOption method with the following parameters:

EndPoint.SetOption(OptSIPPacketEdit, OptTrue);

The next step is create handlers for OnSIPPacketSent/OnSIPPacketReceived events.

This handlers will get “Cookie” parameter. To modify the packet’s content the handler should call  SetSIPPacket with this “Cookie” parameter and new content. Please be aware  - the “Cookie” parameter  is valid only in context of the event handler.