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About Voip Technology : RTP over TCP

hello All

I am reading information about VoIP technology For that i am concentrating on SIP (Session Initiation Protocol) and RTP (Real Time Transport Protocol).

I am interested in implementing RTP over TCP I found that there are some disadvantages of TCP, some are

1) TCP doesn't support multicasting.

2) Through TCP is reliable, it heavily depends on retransmission of

lost or corrupted packets. That increases Delay. Our aim is real

time delivery of packets not reliability.

3) TCP congestion control mechanism decreases the congestion window

when packet losses are detected.

Apart from these flaws i am interested in implementing RTP over TCP

Is it possible ? What is feasibility of implementing RTP over TCP ?

One more point is What is feasibility of implementing RTP over TCP in

case of NAT (Network Address Translation) is there ?

Need your valuable suggestion.

Thank You !

Atul

Reply to
Atul
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It sounds like a bad idea.

Why?? The flaws you mentioned should be sufficient to discourage any attempt of that kind!

It makes no sense: RTP is a layer designed to run on top of unreliable protocols, providing some of the benefist of reliable protocols (e.g., guaranteed ordering, achieved through timestamps) without the drawbacks you mentioned.

If you really want to send multimedia data over TCP, why don't you just use plain TCP? That might work if your application is non-interactive (e.g., A/V broadcasting rather than telephony) because aggressive buffering may smooth over the gaps caused by retransmissions of lost packets. That's what many current Internet media players and servers do in alternative to RTSP+RTP or proprietary equivalent protocols; in fact, they able to handle streaming content over HTTP, which is layered on top of TCP.

Mostly, it will make things worse. There are NAT traversal techniques ("hole punching") that are much easier to implement with UDP than TCP. Even old bad STUN is only for UDP (that's where the 'U' in "STUN" comes from).

If all you need is to encapsulate existing RTP flows (posibly adding security), I suggest you to use some VPN protocol such as PPTP. Or, if you just need to add security to an application, consider using SRTP

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Enzo

Reply to
Enzo Michelangeli

My understanding is that voice runs over UDP and not TCP. UDP does not re-order the packets if they arrive out of sequence hence the need to run RTP which provides re-ordering of packets.

UDP does not have a congestion control mechanism, nor does it send acknowledgements like TCP so there is are no retransmits.

SIP is an alternative to RTP, designed (originally) for voice over the Internet

I could be wrong as I am not an expert and still learning. I'll be interested to hear others comments on this.

Farouq.

Reply to
Farouq

Actually that's not true: SIP (Session Initiation Protocol) is used, as its name says, for the initial establishment of the session, so that the two peers can exchange their IP addresses, the UDP port numbers for the RTP data packets, the codecs that they support etc.

Enzo

Reply to
Enzo Michelangeli

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