Posted by: Mudassir Ali | December 7, 2012

Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP)
SIP is a signaling protocol widely used for voice and video.
Traditionally if your enterprise wants to connect to PSTN could, you would need BRIs, PRIs, PSTN gateways.
With SIP Trunking service you can avoid all the Network PSTN connections.

What is SIP Trunk:
Internet telephony service provider (ITSP) offer SIP trunk as a service for communication between enterprise PBX and PSTN.
IP Phone——-CUCM———–SIP Trunk————-ITSP——–PSTN
Each SIP trunk can have multiple voice session based on enterprise needs.
G.711 – 17 calls over T1
G.729a – 45 calls over T1
SIP trunk is not limited to voice, it can also help enterprise to setup instant messaging, real-time presence, video, etc.
For more details refer to RFC 3261

Benefits of having SIP Trunk:

  1. Like I said no need to invest in PSTN gateway and voice cards.
  2. Off course low cost compared to traditional PRI T1.
  3. Reach out to the world on cost of local call.
  4. Easy installation and maintenance.
  5. Optimal use of bandwidth as data and voice run on same connection.
  6. Move away from T1/E1 capacity limitations of 23/30 channels.
  7. SIP trunk normalization and transparency
  8. Up to 16 destination IP addresses per trunk
  9. Automated hunting from a primary PBX to a secondary
  10. Automated failover to PSTN with full trunk

For you to deploy SIP in your enterprise there are two things which needs to be there.

  1. SIP enabled PBX
  2. SIP enabled edge device

Bandwidth Utilization:
It is always good idea to reserve 27 Kbps with G.729  per call and 84Kbps for G.711

Minimum Bandwidth Requirement
Codec Voice Bit Rate
G.711 64 Kbps
G.729 8 Kbps


Cisco Recommendations:

Application H.323 MGCP SIP Preferred
Voice Mail Control of Individual Ports MGCP
Configuration Dial Peer Based Centralized in CUCM Dial Peer Based MGCP
Load on CUCM Least MGCP
Q.SIG Tunneling Only between PBXs Supported MGCP
Video H.320 ISDN H.320 ISDN H323/SIP
Fax & Modem Pass-through, T.38 Pass-through, T.38 Pass-through, T.38 H323/SIP/MGCP
Port Density High Density Cards MGCP
Redundancy Range of options with dial-peers Range of options with dial-peers H.323/Sip
Security IPSec and SRTP IPSec and SRTP TLS and SRTP SIP
Voice XML Supported Supported H.323/SIP


 Centralized SIP Trunk Design Limitations

MoH Centralized MoH Limited to 50
Central Site Device Pool Multiple Device Pools for Devices on Datacenters
Non Ported DIDs Requires a Different Call Flow and Different Call Routing
FAX Not Supported on SIP Trunk; Handled by Site GW
SRST Limited Access via FXO, PRI for Medium/Large Site
DTMF SIP Trunk and Check IP Phone

Basic SIP Trunk configuration in CUCM

  1. Step 1:
    1. Create a SIP profile (optional).
    2. Create a SIP trunk security profile (optional)
    3. Create a SIP trunk.
    4. Configure the destination address.
    5. Configure the destination port.
  2. Step2: Associate the SIP trunk to a Route Pattern or Route Group.
  3. Step3: Configure SIP timers, counters, and service parameters, if necessary.
  4. Step4: Reset the SIP trunk

Common SIP Requests

  1. REGISTER – to register a phone or line with a SIP Server
  2. INVITE – to set-up a call
  3. CANCEL- to cancel a call set-up
  4. BYE – to terminate a call

Common SIP Responses

  1. 100 trying
  2. 180 ringing
  3. 200 OK
  4. 401 not authorized
  5. 404 destination not found
  6. 486 busy

SIP Trunks vs. H.323 Trunks (Inter cluster)

H.323 SIP
Annex M1 Features / Q.SIG Tunneling NO NO
Signal Authentication No YES
Media Encryption YES YES
GK Support YES NO
SIP Proxy Support NO YES
iLBC and G.Clear Support No YES
G.722 Support YES YES
Multicast MoH YES YES
SIP Subscribe/Notify, Publish-Presence NO YES
Path Replacement NO NO
Call Completion to Busy Subscriber NO NO
Call Completion No Reply NO NO
Message Waiting Indicator (On/ Off) No YES
Alerting Name NO YES


SIP Trunking byChristina Hattingh, Darryl Sladden, ATM Zakaria Swapan Published by Cisco Press.

Cisco Unified Communications System 8.x SRND
BRKUCC-2006 – SIP trunk design and deployment
BRKCCT-2030 – SIP based Architectures for Cisco Contact Center Solutions & Collaboration
BRKUCC-2735 – SIP Trunk Design and Deployment Playbook for the Enterprise
BRKUCC-2450 – Planning for SIP trunking and dial plan centralization with SME



  1. Some good info on SIP Trunking. I have not been so clear when it comes to SIP Trunking but thumbs up for this 🙂

Leave a Reply

Fill in your details below or click an icon to log in: Logo

You are commenting using your account. Log Out / Change )

Twitter picture

You are commenting using your Twitter account. Log Out / Change )

Facebook photo

You are commenting using your Facebook account. Log Out / Change )

Google+ photo

You are commenting using your Google+ account. Log Out / Change )

Connecting to %s


%d bloggers like this: