Lecture Notes: Call Setup and Teardown Process

Introduction

• Understanding call setup and teardown is crucial for network troubleshooting.
• Knowing the process aids in analyzing network packets and identifying faults.

Call Setup Process

Initial Call Setup

• User lifts handset and receives a dial tone from the call manager server (CUCM).
• User dials the number (e.g., 1000 dials 2001).
• Digits are sent to the call control agent (CUCM or voice-enabled router).

Endpoint Discovery

• Call manager identifies the location of the called party.
• Endpoint could be local or require routing to a remote destination.

Permission Check

• Call manager verifies if the calling party has permission to dial the number.

Bandwidth Check (Call Admission Control)

• Determines if there is sufficient bandwidth for the call.
• Could be controlled via Call Admission Control or RSVP.

Call Progress Tones

• Once the endpoint rings, progress tone is sent to the caller.
• Engaged progress tone is played if the called party is busy.

Call Logging

• Once answered, Call Detail Records (CDR) log call details.
• Logging can be done via syslog, radius server, or direct database entry, depending on the call control agent.

Capability Negotiation and RTP

Capability Negotiation

• Determines a common codec between endpoints.
• Failure in negotiation may result in call failure or require additional devices.

RTP and RTCP Negotiation

• Endpoints negotiate ports for RTP (carries conversation) and RTCP (monitors RTP quality).
• RTCP ports are one increment above RTP ports.

Call Teardown

• Call ends when either party hangs up.
• Signal sent to call manager to release resources.
• CDR reports call completion and releases bandwidth for other calls.

SIP Call Flow Example

SIP Call Initiation

• User Agent Client (UAC) sends an invite to User Agent Server (UAS).
• UAS responds with messages (100 Trying, 180 Ringing).

Call Establishment

• Upon acceptance, an ACK is sent, and RTP stream is negotiated.
• Bi-directional communication commences.

Call Termination

• "BYE" message ends the call; acknowledged by a 200 OK response.

Session Description Protocol (SDP)

• SDP describes media session parameters.
• Used for session announcement, invitation, and parameter negotiation.

SDP Parameters

• V: Version
• O: Origin (username, session ID, version)
• S: Session Name
• T: Time (start and stop)
• C: Connection Data (IP address)
• M: Media (type and codec)

Codec Examples

• RTP port numbers indicate codec usage (e.g., G.711, G.729).

Early Offer vs. Delay Offer

• Early Offer: SDP sent with initial invite.
• Delay Offer: SDP sent after initial invite, allows receiver to send first.
• Configuration of SIP trunks can specify early or delay offer.

Configuring SIP Trunks

• SIP Trunks can be configured to enable early or delay offer.
• Media Termination Point (MTP) settings impact offer type.

Conclusion

• Understanding SIP, SDP, and signaling processes are key for troubleshooting VoIP networks.
• Knowledge of codec and RTP negotiation is essential for call setup efficiency.