Which is the best SIP compatible codec type for Android_问答_开发者

I want to develop a Android App which will use a SIP Serve开发者_StackOverflowr of my client. My client is exposing couple of REST API from the SIP server for communicating with the apps.

I want to know which would be the best codec type for this app?

Basically, I want to create a SIP-Stack and send the SIP Packets to the Server. So, there should be a coding and decoding system for the packets. My client prefers 16 kb/sec but I am not sure which should I use.

As others have said, SIP does not transfer audio or video. Although in theory, you can send data over any transport, including ATM, analog lines, a DS0, etc, in the real world, RTP is the most common. RTP (Real Time Protocol) and RTCP (Real Time Control Protocol) or SRTP (Secure RTP) usually carry the audio and video.

As far as codecs go, you will be limited by what your server supports. Here are a few common codecs and some pros and cons of each.

G.711 - Toll quality (ie good as a good analog phone line, or even a bit better). "Universal" in that virtually every device supports G.711. Takes a lot of bandwidth, it doesn't really compress data (G.711 is a "compander"). The baseline G.711 is pretty bare-bones (its really a couple of look up tables). Appendix I adds packet loss concealment (PLC) and Appendix II adds silence suppression and comfort noise generation.

GSM - used on cellphones, sounds ok, good PLC, good compression

G.729A - widely used, near toll quality, good compression (8Kbps)

G.723.1 - widely used, almost as good as G.729, better compression (4-5Kbps)

G.722 - sounds better than G.711, wideband (twice the audio bandwidth of G.711 or an analog call), same bandwidth used on the line as G.711

GIPS - various implemnetations exist, one is free. IIRC, uses about 13.5Kbps on the line, sound is not as good asG.723.1 (but this is a perceptual metric, YMMV) Takes a lot of processor.

All the codecs use some processor and other system resources, as a rule of thumb the more aggressive the codec (the smaller the bandwidth) the more processor used. Also, all of these particular codecs are lossy codecs--they lose some of the data. This means that there is compression, not that portions of the audio are dropped due to poor routing and poor line quality. Much like an MP3 is considered a LOSSY codec while FLAC is considered Lossless. If you're interested the following wikipedia article explains in further detail: http://en.wikipedia.org/wiki/Lossy_compression

You need to know what codecs and protocols this SIP server will support. If you control both ends and want to stick to 16Kbps, you'll want iLBC (no royalties) or G.729 (royalties apply). G.711 and (now) G.722 have no royalties either, but both use ~64Kbps.

The list given before is good, with a few issues.:

GIPS - various implemnetations exist, one is free. IIRC, uses about 13.5Kbps on the line, sound is not as good asG.723.1 (but this is a perceptual metric, YMMV) Takes a lot of processor.

GIPS is not a codec - iLBC and iSAC are codecs designed by GIPS. iLBC is free and standardized. iLBC is high quality, 13 or 15Kbps, and is very resilient to packet loss compared to G.729 or even G.711. You can have 30 or even 50% loss with iLBC and still be understood. I'm not sure I'd say it uses a lot of CPU compared to say G.729.

All the codecs use some processor and other system resources, as a rule of thumb the more aggressive the codec (the smaller the bandwidth) the more processor used. Also, all codecs are lossy codecs--they lose some of the data.

Well, G.711 isn't really lossy per-se (in theory yes, but it's almost more quantization-level loss). 64K G.722 isn't very lossy either. G.723 sucks dead gerbils through garden hoses. :-)

It sounds like a bad idea to do it yourself. Developing a sip client is not a trivial task since there are several protocols you would have to implement. Choosing the coding is not very important decision compared to the rest.

imho you should use one of the open source sip stacks available (like pjsip) or build your application on top of an open source sip client (like sipdroid).

But since you asked for codec: Use the GSM codec. Saved bandwidth and sounds OK. G.711 is otherwise the standard codec that 99% of all sip servers support.

Any?

SIP does not send and deal with ANY data packets. SIP is the Session Initiation Protocol and it handles the NEGOTIATION OF SESSIONS.

The sessions then arae - in case of auio and video - are based on RTP and use RTSP for signalling. So, your question indicates a REAL lack of knowledge of what you need to do - the real uestion is: you need a RTP compatible codec.

Which is similar senseless. RTP is jsut a carrier protocol. THis is like asking "what is a HTTP compatioble image format". HTTP does not care. The browser does.

In case of RTP, it means - RTP does not care. It can transport ANY data. WHat is important is that BOTH SIDES know the same codec. So, in your case it means:

If you program both sides then it is your choice.
If you program only one sidwe (like a SIP phone system), then the question is waht "normal" programs handle.

There are two things you need to take into consideration here:

What other devices/servers that handle media are being deployed or are planned to be deployed
Is your customer looking for narrowband or wideband solution - this will affect the voice quality of the call greatly

Once you have nailed down the answers to the two questions above you will be able to select. For mobile devices, the voice codecs usually used are AMR-NB or AMR-WB. For SIP it is usually G.729 or G.722.x. You also have Speex, ISAC and SILK to choose from. You will probably need to do G.711 in any case just to interoperate with everything - bandwidth will be higher though.

No easy answer here. If your customer can select, or state what other devices are being used - it will be easier for you to select.