RTMFP vs RTMP | Flashphoner

Flashphoner team proudly reports to all Flash and VOIP community that Flashphoner company released two products which may be interested?for VoIP experts as well as for Action Script developers.

Both of that products implements RTMFP protocol.
Please note letter ?F? here in protocol name.
RTMFP suits perfectly for sending audio/video in realtime, because this is UDP-based protocol, unlike its ancestor ? RTMP protocol, which is TCP-based and on which created mostly of the products having similar functionality ? sending audio/video in realtime. That is why we use RTMFP as main protocol in our products, and got really impressive results.

In the beginning we are working a lot with Flash Player and Wowza Media company proucts. And all working greatly on synthetic tests, but in real production we`ve got a problem ? that was?a growing latency of aaudio and video. And we realized that it can`t be resolved by the reason of?TCP protocol (on which RTMP is based) where each packet must be delivered to adresses inthe strict order. This is called a ?reliable protocol?. And for that reliability we need to pay by growing?latency, which is unacceptable for professional VOIP solutions.

What is the latency?
This is when one user says the word, and the other person hears that word only after N seconds. During the tests with RTMP delay could be up to 4 seconds for an average network quality, and this is really kills the quality of communication. All the professional VoIP softphones sends audio/video via UDP protocol to minimize the latency. And latency was the last line before creating?full featured softphone on Adobe Flash Player, as it already have all other needed features.
That is: ability capture, publish and playback audio/video, support mainstream codecs (h264, g711, Speex wideband),echo cancellation, jitter buffer.

Now about the products.

Flashphoner RTMFP SIP Gateway

This is the gateway between Flash Player and SIP environment.?It allows you connect to any SIP servers (Asterisk, OpenSIPs etc.), make and receive calls from PSTN/mobile from your flash-based?softphone from the browser.
Yes, really ? you will not only be able to call from the Click-to-Call widget?to mobile phone, but you can also call FROM mobile to the user on the website! This is really impressive.
You can also send instant messages / SMS here. If your SIP server/provider supports sending and receiving SMS, you can send SMS from the browser to mobile phone, and even receive sms from the mobile phoneto your browser softphone. Usually for that used full-features IMS (Internet Multimedia System)?working on 3GPP standards.

Flashphoner RTMFP Media Server

This is Media Server, intended for creating simple Flash-to-Flash videochats or videoconferences. Today there are many Flash-based videochat platforms based on RTMP protocol and all of them suffers by growing latency when one of the user has a bad network connection. If you go to the forum of any manufacturer of RTMP chat ? you will find many topics about latency, which is sadly finished by something like ?I still have a latency ??

Flashphoner RTMFP Media Server solves that problem and make Flash videochats professional solution (based on UDP (RTMFP) protocol) for sending audio/video realtime. If you have a latency of a video traffic ? you can still talk and understand each other, but if there is audio latency ? you can`t talk at all. You just will not understand each other.

Please note, that currently we are not providing any Peer-to-Peer solutions. Some time ago we are working with Peer-to-Peer technology based on Flash RTMFP, but quickly realized that Peer-to-Peer connection not works in mostly cases. We had pilot project like Chatroulette and we got number of 10% successfull Peer-to-Peer connections. Other ones was connected through the server. That happens becuase there are many NATs and Firewalls in the global worldwide, which often do not allows to establish Peer-to-Peer connect. So at the moment Flashphoner RTMFP Media Server will not allows you using Peer-to-Peer connections. But if you need professional Flash-based videochat which works through the server ? Flashphoner RTMFP Media Server is what you need.

Another nice feauter of videochat, based on Flashphoner RTMFP Media Server?is the ability of extending and integration. We use Java on the server side for implementing?server processors. That mean you can use all the power of Java platform for integrating with the databases, third-party web-services, protocols and other.

Specifications

Flashphoner RTMFP SIP Gateway

- Codecs ? G.711, G.729, Speex Wideband, H.263, H.264, Sorenson Spark
- Protocols - RTMFP, SIP (RFC3261), RTP
- Instant messaging ? RFC 3428
- Transfer, Hold ? RFC3515
- Call recording
- Javascript API
- Client application ? Javascript Phone (open source, ready for customization)

Flashphoner RTMFP Media Server

- Protocol ? \t RTMFP
- Codecs ? G.711, Speex Wideband, Nelly Moser, H.264, Sorenson Spark
- Commands ? connect, disconnect, publish, unpublish, play, stop
- Extending - Ability to use Java on the server side, invokes customed methods from server to clietn and back

About a tests

As we told above, RTMP (TCP-based) server will try to keep ALL of packets always and by any price,?even it will be erious snetworks lags. That make latency growing, because server will wait?every missed packet before playing the next. More lags ? bigger latency. If we will try drop the missed packets programmatically ? we will get big percent of loss in the sound, which is extremely bigger then real loss rate of the network. This is confirmed by tests with synthetic increasing of the ?loss percent? by using??Linux netem? and ?IFB? utilities.

At the same networks conditions (with the same lags), RTMFP works without latency growing and with little sound quality deterioration. And that deterioration is much more less compared to RTMP case, where server drops really a lot of incoming TCP packets to minimize the delay. This difference is especially evident on the sound stream, where one UDP packet?(when RTMFP used) has one audio frame, and loss one or even few of them is not?a really threat to the overall quality of communications.

Things are a bit different when you are using video. Some UDP packet carries a video key?frames with the resolution of 640?480, for instance. In the case of loss one or more such packets you can have ?artefacts? during the decoding. But you still will have low latency. And when lags will go on ? picture will recover.

Also there are special parameter of the NetStream API in the Adobe Flash Player, which controls reliability of the UDP packets delivery. That parameters specifies separated for audio, video and data. Reliability is the most difference between UDP and TCP. When you?set reliability ?true?, UDP protocol becomes similar to TCP. That mean you can control how?to delivery different kinds of content. For example you must always set reliability=true for data delivering. Because ?data? content carries all of SIP siganiling information, and you can not lose any piece of that ? it conatains instant messages, and all the call control information. For the audio and video you can use ?true??or ?false? depends on network quality and your requirements to the quality of sound and picture.?You can set audioReliable=false, and videoReliable=true simultaneously and you will?have acceptable quality of both of them in the most networks.

Thus, based on the results of tests, we can say that RTMFP protocol significantly improves latency/quality ratio of a Flash VoIP communications, and that protocol?can be recommended as a platform for developing web-based applications that?provide realtime audio/video communication services with the low-latency and?acceptable voice quality.

Hope that we could briefly bring information about the presented products and their capabilities. We suppose, you have some questions and suggestions about it.
We are welcome your feedback.?Please write us to [email?protected]

Thanks!

Source: http://flashphoner.com/rtmfp-vs-rtmp

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