What's the Difference Between VoIP and SIP?

What’s the Difference Between VoIP and SIP?

SIP and VoIP are terms that can often be interchanged, but not always. Let’s discuss the differences between the two.

What is VoIP?

VoIP stands for “Voice over Internet Protocol”. Simply put, VoIP covers any type of voice communication that uses TCP/IP networks to pass the audio between devices. (TCP/IP is the layer 3 protocol that makes up the bulk of LAN and WAN/Internet traffic) Examples of VoIP include IP phone calls, Zoom meeting audio, web conferences with audio, remote radio operating, etc.

There are different layer 7 protocols all classified as VoIP such as:

• SIP (the most common)
• H323 (an older, less used standard)
• IAX (the asterisk protocol)
• Skinny Client Control Protocol (SCCP, the Cisco protocol)
• UNISTIM (the Nortel protocol)

The actual audio in VoIP is carried using a CODEC which turn sound into bits & bytes. The most common codecs are:

• uLaw (N.A. standard)
• aLaw (European standard)
• G729 (compressed)
• G722 (high fidelity)
• OPUS

What is SIP?

SIP stands for “Session Initiated Protocol” and is a subcategory of VoIP. The SIP standard dictates what messages should look like when setting up a phone call between devices. For instance, a SIP call always begins with an INVITE message. The typical response is TRYING, followed by OK if the call is completed. The sending side would then send an ACK. The INVITE and OK messages contain something called SDP – Session Description Protocol. The SDP portion tells the two sides which codec to use and what other actions are allowed.

Here is an example of some SIP messages:

INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.125.1.145:4999;branch=z9hG4bK2897671491
From: "Office Cordless" <sip:[email protected]:5060>;tag=3108188849
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:4999>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink W60B 77.81.0.35
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 282

v=0
o=- 20043 20043 IN IP4 10.125.1.145
s=SDP data
c=IN IP4 10.125.1.145
t=0 0
m=audio 12130 RTP/AVP 0 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

----------------------------

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.125.1.145:4999;branch=z9hG4bK3624976775;received=85.140.52.21
To: <sip:[email protected]:5060>;tag=4nXKByldsEV4ydDI6DA7DC
From: "Office Cordless" <sip:[email protected]:5060>;tag=3108188849
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:85.140.52.22:5060;transport=udp>
Server: NetSapiens SiPBx v41-2-4
Content-Type: application/sdp
Content-Length: 240

v=0
o=NetSapiens_Nms 1655755868 1655755868110 IN IP4 85.140.52.22
s=SIP Media Description
c=IN IP4 85.140.52.22
t=0 0
m=audio 28656 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Still Got Questions?

Is there a detail this post didn’t cover? Feel free to post your question on this VoIP Technical Forum.