VoIP

Wowza Streaming Engine
21

sounds interesting… i am also looking for a way to connect SIP voIP with Flash Media Server.

so maybe i can be of any help!

greetz,

Mart

22

I will send email to a developer that might be able to help, and alert him to this post.

Richard

23

Mediaman,

How can we connect?

The work is divided in two major tasks:

  • Provide a real time media sink between FSM and Sip Server.

  • Establish the SIP connection. For this we have found good alternatives.

We are starting the project, so if you can join along that would be great.

gcaeiro and mediaman: I´ve been also assigned the task of investigating about FMIS sending streams thrugh a SIP server. I am just starting up with my investigation, we´re a group of 3 actionscript programmers. If there´s any advance you would like to join with us and any tasdk you think we could work on in order to help this development happen, just let me know!

24

Hi - we are also looking for a Flash front end to our video SIP server ASAP. I can be contacted on wrn1976 at gmail dot com - please get in touch if you’re working on a project (or know of one) as we’d love to be involved.

25

Hey guys,

did you get any step further?

i try to write a server application that connects to asterisk, allowing multiple flash clients to make and receive phone calls. so i’m struggeling with the same probs:

a) transcoding the nellymoser in real-time to an apropriate codec, so converting the rtpm stream to a voip rtp stream, this has to happen vice versa which means to create a flash stream from an incoming asterisk SIP-call

besides that I’m no very familiar with the server side api so java. how can i grap the stream…maybe with “onstreamcreate()”? and howto convert it in real-time. there are some progs out there e.g. FFmpeg or nelly2pcm. i saw that FFmpeg is able to convert in real-time using some obscuring pipe commands…

b) the next task is to initiate the SIP call but this is probably more a matter of study, getting familiar with the java sip protocol stack or using some open source java soft phone

robroy

26

May be this is an old post. But does this mean that raw access to video and audio data is available from an rtmp stream ?

27

I am also looking into this problem of making a SIP audio call with an existing RTMP stream (live/vod). I can integrate with nelly2pcm or ffmpeg to re-stream it. Did anyone make any progress on this? I would like to contribute in any way I can.

28

Charlie,

Can you be a little bit more specific as to what fetures will enable this kind of applications ? It would help us a lot to plan and think through

Thanks

Siva.

29

I am not sure how much speex will help although there might be quite a few speex based installations.

RTP out might be an interesting prospect, but what are your thoughts on transcoding (or some kind of a plug-in infrastructure if ppl choose to plugin their transcoders in the path) ?

Also, did anyone succesfully use the raw stream access module for this kind of applications?

30

Hi,

i wonder if ther are now an easy solution to broadcast audio from an sip provider to wowza server ?

Because sometimes the bandwith quality is very low or users don’t have microphone on their computer, we are looking for a solution to broadcast audio from a standard phone calling an sip account connected to a wowza application.

thanks for your help,

regards,

Jerome.

31

thank you Charlie.

but i’m looking for something more detailed about connecting sip account to wowza server…

any experience?

regards,

Jerome.

32

Actually my request is easier. I would like to set up a one way feed from VOIP --> Wowza and then record on the server.

I am currently thinking

POTS --> SIP --> Asterisk --> VLC (mp3?/NellyM) --> RTP --> Wowza --> flash player

has anyone done this?

33

Hi,

I am trying to achieve wowza (RTMP)-> RTP conversion so that I may connect to any standard sip/323 servers.

I am quite late in the race… have you guys been successful at it? what does wowza provide to make it possible ?

In one of the threads here https://www.wowza.com/forums/showthread.php?t=142, Charlie mentions that Wowza provides deeper APIs than FMS to develop VoIP apps. Is it still true? I am interested in getting something out by march end. Can we work together ?

Feel free to contact me at my login id at gmail

Thanks

Srini

34

Any body who have developed VOIP application using Wowza? Any known performance issues?

35

Hello ,

has anyone implemented a flash to sip transcoder using wowza? If yes then please contact with me. We are willing to buy an rtmp to flash transcoder.

Best Regards

36

Hello again ,

forgot to mention that we need support for video calls with H263 codec.

37

panpaterakis

Hello, my name is Pavel!

Our team developing product exactly what you need.

Please take a look here http://flashphoner.tooboos.com

Old name of this product is WSPlugin.

At the beginning of next week will be updated Flashphoner website

and released the latest version of the product with documentation.

Flashphoner will have month trial version for 10 connects, so you could test it.

Flashphoner - Flash-to-SIP gateway based on Wowza Media Server 2

Flashphoner support voice calls on (Flash Player + Wowza Media Server 2 + VoIP server) platform for SIP/RTP protocols.

It make possible to develope SIP phone systems based on Adobe Flash Player

and Wowza Media Server 2.

Specification

  • Supports streams in Speex 16kHz (Adobe Flash Player 10).

  • Resampling Speex 16kHz <-> Speex 8kHz for compatibility with VoIP devices.

  • Transcoding Speex <-> G.711 codec.

  • SIP implementation between Flashphoner and VoIP server via SIP 2.0.

  • Exchange RTP/G.711 - streams between Flashphoner and VoIP server.

Requirements

  • Linux

  • Adobe Flash Player 10

  • Wowza Media Server 2

  • JDK 1.6

  • gcc (GNU compiler collection required for Linux installation)

Hardware requirements

  • Intel Pentium 4 with a 3 GHz processor(dual-core Intel Xeon recommended)

  • 1Gb RAM

  • Ethernet Card

Please write us your contact and we send you last release with documentation when it will be done. Also this version will be released on Flashphoner website.

38

panpaterakis

Supports of H263 codec will be added during next two weeks

39

maybe it’s a bit a too late but I’m confused after reading different posts, what we want to have is that some users can have access to a conference simply using telephone, beside the normal users. So, what we thought is, first selecting the audio signal from Wowza server, then with VoIP server establishing calls. I appreciate any hints or code examples.