[Asterisk-Users] 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 Thosar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Voip Technology : RTP over TCP
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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Voip Technology : RTP over TCP
On 5/13/05, Atul Thosar [EMAIL PROTECTED] wrote: 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. TCP stacks don't always support the standard either.. 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. quite the opisite. reliably cheaper calls... 3) TCP congestion control mechanism decreases the congestion window when packet losses are detected. yes and your point... Apart from these flaws i am interested in implementing RTP over TCP read about beowulfs and the use of interconnects over Ethernet.. Is it possible ? yes, been done What is feasibility of implementing RTP over TCP ? see above One more point is What is feasibility of implementing RTP over TCP in case of NAT (Network Address Translation) is there ? what port? DNAT? SNAT? loaded question. Need your valuable suggestion. Thank You ! Atul Thosar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About Voip Technology : RTP over TCP
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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
