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VoLTE L3 Analysis

The document discusses signaling protocols like SS7 and SIP used for setting up voice calls over cellular networks. It describes how SIP works with other protocols like SDP, RTP and SRTP to carry multimedia sessions. The document also lists SIP response codes and analyzes VoLTE call flows and quality metrics like jitter, packet loss and delay.

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Haziq Naveed
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29 views12 pages

VoLTE L3 Analysis

The document discusses signaling protocols like SS7 and SIP used for setting up voice calls over cellular networks. It describes how SIP works with other protocols like SDP, RTP and SRTP to carry multimedia sessions. The document also lists SIP response codes and analyzes VoLTE call flows and quality metrics like jitter, packet loss and delay.

Uploaded by

Haziq Naveed
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
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VoLTE L3 Analysis

Signaling System 7 (SS7)


The set of SS7 telephony signaling protocols is responsible for setting up and terminating
telephone calls over a digital signaling network to enable wireless cellular and wired
connectivity. It is used to initiate most of the world’s public telephone calls over PSTN (Public
Switched Telephone Network)
SS7 is not particularly suitable for carrying packet data.

Session Initiation Protocol (SIP):


SIP was originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler and
Jonathan Rosenberg in 1996. The Session Initiation Protocol (SIP) is a signaling protocol
used for initiating, maintaining, and terminating real-time sessions that include voice, video
and messaging applications.[1] SIP is used for signaling and controlling multimedia
communication sessions in applications of Internet telephony for voice and video calls, in
private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as
well as mobile phone calling over LTE (VoLTE).
SIP works in conjunction with several other protocols that specify and carry the session
media. Most commonly, media type and parameter negotiation and media setup are
performed with the Session Description Protocol (SDP), which is carried as payload in SIP
messages. SIP is designed to be independent of the underlying transport layer protocol, and
can be used with the User Datagram Protocol (UDP), the Transmission Control Protocol
(TCP), and the Stream Control Transmission Protocol (SCTP). For secure transmissions of
SIP messages over insecure network links, the protocol may be encrypted with Transport
Layer Security (TLS). For the transmission of media streams (voice, video) the SDP payload
carried in SIP messages typically employs the Real-time Transport Protocol (RTP) or the
Secure Real-time Transport Protocol (SRTP).

IMS
SIP CAUSE CODES

1xx – Provisional Response


• 100 (Trying)
• 180 (Ringing)
• 181 (Call Is Being Forwarded)
• 183 (Session Progress)

2xx – Successful Responses


• 200 (OK)

3xx – Redirection Responses


• 302 (Moved Temporarily)
4xx – Client Failure Responses
• 401 (Unauthorized)
• 403 (Forbidden)
• 408 (Request Timeout)
• 480 (Temporarily Unavailable)
• 481 (Call/Transaction Does Not Exist)
• 486 (Busy Here)
• 487 (Request Terminated)

5xx – Server Failure Responses


• 500 (Internal Server Error)
• 502 (Bad Gateway)
• 503 (Service Unavailable)

6xx – Global Failure Responses


• 600 (Busy Everywhere)
• 603 (Decline)

VoLTE Call Flow


QCI Table

Jitter

Is a variation in packet transit delay caused by queuing, contention and serialization effects
on the path through the network. In a live network, the jitter 98 percentile is below 60ms to
89ms. There is no clear correlation between Jitter and MOS, when the jitter is not higher
than 80ms MOS is good.
PACKET LOSS

RTP PACKET DELAY


VoLTE Accessibility Analysis Flow
VoLTE Retainability Analysis Flow

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