You pick up a handset, dial a number, it rings, and someone answers. How hard can it be?
Actually, a lot goes on behind the scenes with two protocols set into motion—SIP and RTP—to get VoIP to work in a modern network.
For comparison, when you enter www.google.com into a browser, there are two protocols at work: DNS and HTTP. The browser first does a DNS lookup to learn the IP address for the given www.google.com domain name. Once the IP address is learned, it opens an HTTP (or HTTPS) connection to the IP address and begins to download the web page.
A similar mechanism was developed for VoIP where there are two protocols that do the heavy lifting: SIP and RTP.
SIP is the control protocol, and RTP is the payload protocol used to send and receive the voice audio stream.
SIP Ladder
How Does SIP Work?
Session Initiation Protocol (SIP) is designed to handle the "administrative" part of managing a phone call. It will look up IP addresses for given phone numbers, determine if the phone is available, ring the phone, and start and stop RTP streams.
Is SIP Affected by Latency?
SIP packets are administrative in nature and thus are not affected by latency.
Is SIP Affected by Jitter?
SIP packets are not affected by jitter.
Is SIP Affected by Packet Loss?
SIP does have the ability to re-transmit lost packets, but if too many are dropped a call will not be able to be established.
How Does RTP Work?
Real Time Protocol (RTP) carries the voice payload across the network from transmitter to receiver. This payload is a continuous stream of packets that traverses the network.
Most calls involve two streams: One for each endpoint. Thus, each endpoint is transmitting a stream of packets to the remote endpoint as well as receiving a stream from the remote endpoint.
Is RTP Affected by Latency?
Yes, RTP packets are affected by latency. Refer to the blog entry on VoIP Latency for more details on this.
Is RTP Affected by Jitter?
Yes, RTP packets are affected by Jitter. Refer to the blog entry on VoIP Jitter for more details on this.
Is RTP Affected by Packet Loss?
Yes, RTP suffers significant quality issues when packet loss occurs. Refer to Packet Loss Testing for more details on this.
VoIP call quality problems can be prevented if the right information is brought to bear about your network's performance.
Review our white paper or contact us with questions about how PathSolutions can solve call quality and many other VoIP issues.
