Type: Posts; User: poktomi; Keyword(s):
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Have you managed to finish the plugin, and would you be willing to share the code? Reason I ask is I am still in search of a way to stream audio in and out of a Teamspeak client.
"My" audio would...
you seem to be barking up the same tree as me. Please read the thread
Streaming to a TS channel on server to see how far (or not) I have got so far.
I will definitely proceed with...
any news on this? And will this be part of the SDK (not compatible with TS3 servers) or the client plugin SDK (compatible)?
This (the different compatibilities of the different SDKs is confusing.....
ok, finally an answer that I understand - even though it's not what wanted to hear . Will there ever be a SDK for the "normal" TS3 server?
The question remains then: how did Phonecaster do it (PSTN - via Asterisk I believe - to Teamspeak gateway)???
ok, and how can I deploy the software that is being developed with the SDK to a Teamspeak production environment?
Otherwise it would mean we are coding purely in SDK space with no chance of ever...
so what do you do? I thought the SDK doesn't work with the current TS3 (kinda silly btw.; are you reading this TS team?)
the question is: we are developing stuff with this SDK and how will this...
At least that is what I am thinking (talking about). I will also be doing this on a Linux server, so let's stay in touch on this one.
And: where in the client_plugin_sdk is a GUI-client. Can't seem...
const *float data parameter
is what format? I need it in PCM signed linear, which should be short rather than float, so I can convert it to G.711.
I do assume this is realtime, so if I...
yo guy, now you got me hooked. pls. elaborate just a little bit.
can you the 5-10 surrounding lines of code too pls?
and i assume this means you are using the snd-aloop "trick" right?
you are right of course, sorry for the typo. However I believe the module to be called snd-aloop.
In fact what seems to be more professional but also probably to high level and complex for what I...
pls. let me know if you find anything. I am (technically) trying to do the same, except that I want to connect audio streams coming out of SIP dialogs (e.g. G.711 encoded audio data from...
0.0.0.0 means to bind to all IP interfaces. So if you have more than one interface the server will bind to all of them.
This is standard parameter usage (at least in Unix).
From what research I have done it could only be done through some kind of "virtual audio device", which unfortunately does not exist for Linux.
I had high hopes on the new SDK, but they offer...
how do I grab the audio data using the SDK and will it be in signed linear or some codec (e.g. Speex, GSM)?
And of course also: how do I push externally provided (e.g. SIP, asterisk) audio data...
Does the SDK allow me to connect many users/identities from one machine.
I need this to build a SIP to Teamspeak bridge. Any help or advice is appreciated.
how would I go about to achieve that?
I can see two approaches:
- extract the encoded audio (Speex) in the hope it is in RTP format from TS
- work on the basis of a virtual audio...
is there any way we can get in direct contact (e.g. can I send you a private message?) That would probably get me started quite fast.
You will find that only my first questions seem...
First of all thanks for the detailed reply.
Regarding the IDE: When will that release (VisStd05) be out and where will it be announced?
In your experience: what's better to develop on Linux/debian...
I'm a newbie to development on TS. My background is in "traditional" SIP-based VoIP development.
I would like to integrate Teamspeak with an existing SIP platform and have started by...
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