SIP to SIP Call recording - android

How can we record SIP to SIP call in android? how to get downlink and uplink access in android using native code?

Take a look on CSipSimple. It is an open source project that has the features you are looking for. Generally it uses a pjsip SIP stack that runs on native code. You can ask question in their mailing list. It is very active.

Assuming you wrote your own Android NDK SIP stack, recording SIP calls on Android is pretty simple. Just combine inbound and outbound audio (MUX), then write combined audio to file system or some place you want to save the recorded calls.

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Does Android native sip stack support video over sip?

I have implemented successfully sip audio call using android native sip stack. But some how i dont see any information regardind video call over sip using android native sip stack. May be they do not support sip over rtp? . Any information would be highly appreciated. i coudlnt find any clear information on stackoverflow regarding this issue.
Android native sip stack has a very limited feature and i wont recommend to any one. It does not support video feature by default. For video feature it will require bidirectional information flow ,which require RTP. So in short answer no. Better try other opensource sip like pjsip.

Using Restcomm SIP & Webrtc without the platform

I am looking for a way to use SIP as signalling protocol for Webrtc in Android. I saw the RestComm open source code and I think it fits my needs. But I already have the infrastructure setup for TURN and SIP server. We use FreeSwitch for that purpose. My Question is, is it possible to use the restcomm android sdk with a infrastructure that is not from restcomm platform? Will it work?
Is there any other library or way that could be used for the purpose? Infrastructure is fixed and couldn't be changed. I need webrtc with SIP or SIP over websocket as the signalling method.
Restcomm Android SDK essentially offers VoIP functionality using SIP for signaling and WebRTC for media, so you shouldn't have any issues integrating with different server components.
Notice though that so far it's tested with Restcomm platform for the most part.
For more information on how to do that you can check Quickstart Guide and also refer to the code for Hello World and Olympus Apps

Android APIs to Auto answer the call and read DTMF tones

I would like to know the following whether possible with Android or not. I keep searching on line but no luck. I feel that may be possible with new versions of the Android. I want check with experts on Stackoverflow.
Shall I Auto answer the call and play a pre-defined Audio file? At this time, we should not use Micro phone or Speaker.
Once we Auto answer the Call, shall we play pre-defined Audio files based on the DTMF tones received from other end? Do we have Native APIs to read and the DTMF tones? - Simply, can I build a "IVR System" as an Android App? (asking too much? send me your suggestions)
Shall we record the telephony streams as Audio files? We can write a trans-coder if we have access to the streams using the native APIs.
I may be asking too much here, because am a new to android and did not find any absolute answer online.
Thanks in Advance,
- PC Varma

How to clearly identify voice codec used in a VOIP application?

I was going through some sample VOIP SDK for Android. The SDK providers say that they are using the G729 voice codec in the SDK. But those codecs are hidden. Is there any way to clearly identify the voice codec used in the application?
There is no way I know if you have access to the raw audio data only. However, if you can somehow access the signaling data (e.g SIP), you can look at the selected SDP payload. If it is 18 then the call is encoded in G.729.
Hope that helps...

Android configure and create sip calls programmatically

I am new to android and SIP. I want to create an application in android through which i can configure and create SIP calls programmatically. I want to encrypt that call.
Android includes a full SIP protocol stack and integrated call management services but it will not manage sessions, transport-level communication.
So anybody who has good idea on SIP/Voip please kindly suggest me how can i achieve this.
Look at CSipSimple program. It is an open source project.

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