[Asterisk-Users] About Voip Technology : RTP over TCP

[Michael D Schelin]

TCP is too slow for Real time Apps.  If you have packet errors TCP will 
try to resend the packet. This will create latency issues.  This is why 
UDP is used for Voip. 1 or 2 missing packets is not going to be missed. 
If you look at your Stats. you'll see a few of them. 

Stewart Nelson wrote:

>>I am interested in implementing RTP over TCP
>>    
>>
>
>Why?  If you want to permit operation through a firewall
>that blocks UDP, there are packages that provide VPN
>tunnels over TCP or even HTTP.  You could then run
>any VoIP system over that VPN.  As you said, delay
>performance would sometimes be awful.
>
>Skype will automatically fall back to TCP if a UDP
>connection attempt fails.
>
>Most of the commercial instant messaging packages
>that support voice or video can work over TCP.
>
>If your purpose is to improve performance on networks
>with high packet loss rates, IMHO you would get better
>results from a UDP-based system that permits forward
>error correction, by transmitting each voice frame
>in two or more packets.  If you can't afford the
>increased bandwidth, a system of retransmission such
>as used by popular streaming protocols would still be
>better than TCP.
>
>  
>
>>One more point is What is feasibility of implementing
>>RTP over TCP in  case of NAT (Network Address
>>Translation) is there ?
>>    
>>
>
>Any of the above systems can work through NAT.  If
>both endpoints are behind NATs, and you can't
>set up port forwarding on either, then of course you
>must connect via an intermediate server.  Skype
>and the IM services do that automatically.
>
>If your desire for TCP is not related to firewalls
>or packet loss, I'd be interested in hearing about
>your application.
>
>--Stewart
>
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