Comcast SIP Operations

June 24th, 2015

Evolution of the Comcast Voice Network

Acquired circuit switched telephone network as part of the ATT Broadband merger
Transitioned to Voice over IP based on PacketCable 1.0/1.5 using CMS islands
Largely MGCP protocol based network
100% of PSTN connectivity TDM based
CMS to CMS calls go off network
Implemented SIP Routing and Peering
CMS to CMS calls stay on Comcast IP network
Begin sending traffic to PSTN via SIP Peering and Least Cost Routing
Now over 80% of PSTN traffic is exchanged via SIP
Launched Commercial Voice solution using SIP Clients for Small Business
Launched ISDN/PRI PBX solution using IADs with SIP Registration
Transitioning to IMS network based on PacketCable 2.0
Geo redundant solution supporting SIP Client registration to any core
Migration of residential subscribers to IMS to be complete mid year
Migration of commercial small business subscribers beginning
Launched support for SIP PBX
Launched Hosted PBX solution with hand/desk sets registration via SIP
Deployed SIP Soft Client solutions for both residential and commercial voice
Mobile OTT Applications

Comcast Voice Network Scale

IMS Network
4 core locations with 2 core instances each
11M residential lines
Commercial
15 Application Servers, 6 Network Servers, and 12 Media Servers
750K commercial lines
SIP Peering and Least Cost Routing
133 Session Border Controllers
20 Direct SIP Peers
12 Least Cost Routing Providers
56 Wholesale Customers
Operator Services and Directory Assistance via SIP
Toll Free via 3 SIP Peers
Commercial Network
4 Application Servers for Small Business with over 250K lines
3 Application Servers for PRI Trunking with over 2M TNs
1 Application Server for SIP Trunking with over 20K TNs
2 Application Servers for Hosted PBX with over 200K Seats
2 Network Servers, 10 Media Servers, 12 Access SBCs, 8 Peering SBCs

Other SIP and WebRTC

SIP Publish
Leveraged for collection of voice quality metrics from all clients
SIP Subscribe Notify
Leveraged for Voicemail Message Waiting Indicator
IMS Registration Event controls
Initially used for CallerID to TV on the X1 platform
First WebRTC Deployment
Xfinity Share Recorded or Streaming personal content delivered to friends and
family X1 Set Top Box
Unified Communications
Enterprise Telecom broadly deployed SIP PBX, Call Manager, and Session Manager
solution
Connected to the broader Xfinity and Business Voice networks through SBC
Audio/Video Conference, Instant Messaging, Collaboration
Support for multiple CODECs including Transcoding on Peering

SIP Performance Management

Complex Voice over IP networks require multiple messages to complete a single call
creating a high volume of SIP and other protocol traffic
Maintaining network health requires more attention to performance management than
fault management (although fault is still important!)
Key Performance Indicators
Sessions and Calls Per Second
Invite, Bye, Cancel, Option, Update, Refer, Ack/Prack Messages
Flow Messages (18X and 2XX)
Error Messages (4XX, 5XX, 6XX)
KPIs can be reported directly from network elements or generated as part of passive
signal capture and analysis
Trends need to be analyzed with threshold alerts applied
Challenge is identifying which KPIs need thresholds and where to set them
Growth is also a challenge along with other impacts to baselines for trending

Sample Dashboard

Whats Next?

Continued evolution of Business Class Feature Set

Implement Border Gateway Control Function for IMS
Ramp up SIP security
New holes to close on Mobile Apps leveraging 3rd party Internet
Deep dive on SIP/TCP fragmentation issues
Fraud detection and mitigation
Virtualization
Signaling in the Cloud
Bearer Traffic??
Voice over IPv6
Big Data Analysis and Correlations
Interop, Interop, Interop!!!

Dial Around Testing

Challenges
How to validate the resolution of a carrier Trouble
Ticket post re-route
How to test a new turn up during ORT
ASR testing pre LCR update

ENUM Query

Call Flow

IRDB

SBC

CMS/IMS

SRP

Peer
SBC

Abbreviations & Acronyms

APOP Application Point Of Presence
ENUM Electronic Number

SAG Session Agent Group

SBC Session Boarder Controller
IRDB Intelligent Routing Dbase
SRP SIP Routing Proxy
ORT Operation Readiness Testing

The User will dial 101+cic+number

In our example the user dialed

101402215036985129

Invite sent to the Proxy

INVITE sip:5036985129;npdi;rn=5036583783;cic=4022@snjscabad01aa.ssg.comcast.net:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP SLKDUTQSPS0aa.ssg.comcast.net:5060;branch=z9hG4bK_1115101141894840168
From: "RTS 1 Port 14"<sip:4352490949@SLKDUTQSPS0aa.ssg.comcast.net;user=phone>;tag=1_1115_f10114_k619_CtkM94DCAE
To: <sip:5036985129@snjscabad01aa.ssg.comcast.net;user=phone>
Call-ID: 1284747884@SLKDUTQSPS0aa.ssg.comcast.net
CSeq: 1

INVITE

Max-Forwards: 70
Supported: 100rel,timer,replaces,unknown
Contact: <sip:4352490949;tgrp=7777;trunk-context=SLKDUTQSPS0aa.ssg.comcast.net@SLKDUTQSPS0aa.ssg.comcast.net:5060;user=phone; >

Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,REFER,UPDATE
Min-SE: 900
Session-Expires: 1800;refresher=uac
P-Asserted-Identity: "RTS 1 Port 14"<sip:4352490949@SLKDUTQSPS0aa.ssg.comcast.net;user=phone>
User-Agent: BTS10200/900-06.00.03.V00 (SIA)
Content-Length:

435

Content-Type: application/sdp

Invite after the In-manipulation HMR

INVITE sip:5036985129;npdi;rn=5036583783;cic=4022@7777.srp.comcast.net:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP SLKDUTQSPS0aa.ssg.comcast.net:5060;branch=z9hG4bK_1115101141894840168
From: "RTS 1 Port 14"<sip:4352490949@SLKDUTQSPS0aa.ssg.comcast.net;user=phone>;tag=1_1115_f10114_k619_CtkM94DCAE
To: <sip:5036985129@snjscabad01aa.ssg.comcast.net;user=phone>
Call-ID: 1284747884@SLKDUTQSPS0aa.ssg.comcast.net
CSeq: 1

INVITE

Max-Forwards: 70
Supported: 100rel,timer,replaces,unknown
Contact: <sip:4352490949;tgrp=7777;trunk-context=SLKDUTQSPS0aa.ssg.comcast.net@SLKDUTQSPS0aa.ssg.comcast.net:5060;user=phone>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,REFER,UPDATE
Min-SE: 900
Session-Expires: 1800;refresher=uac
P-Asserted-Identity: "RTS 1 Port 14"<sip:4352490949@SLKDUTQSPS0aa.ssg.comcast.net;user=phone>
User-Agent: BTS10200/900-06.00.03.V00 (SIA)
Content-Length:

435

Content-Type: application/sdp

SRP config
from-address
*
to-address
7777.srp.comcast.net
source-realm
*
description
activate-time
N/A
deactivate-time
N/A
state
enabled
policy-priority
none
next-hop
enum:e164.TEST; key=cic
realm
siprxy01
action
replace-uri
terminate-recursion
enabled
carrier
start-time
0000
end-time
2400
days-of-week
U-S
cost
10

Enum Query
SNJ-SPRXY-02# show enum lookup e164.TEST +4022

ENUM Lookup Result:

Query Name -->
+4022
Answers -->
sip:+4022;tgrp=PEERXYZ;trunk-context=PRI.SBC@WDSTGAHQ-INT.SBC.SIP.SAG ttl= 60

SNJ-SPRXY-02#

Target SAG within the SRP

group-name

description
state
app-protocol
strategy

WDSTGAHQ-INT.SBC.SIP.SAG

Peer-XYZ Woodstock 9200 SBC SAG

enabled
SIP
RoundRobin

dest
WDSTGAEMBSCaa.ssg.comcast.net
WDSTGAEMBSEaa.ssg.comcast.net

WDSTGAEMBSKaa.ssg.comcast.net
trunk-group
sag-recursion
stop-sag-recurse

disabled
401,407

INVITE sent to the SBC

INVITE sip:5036985129;tgrp=XYZ;trunk-context=PRI.SBC;npdi;rn=5036583783;cic=4022@WDSTGAEMBSEaa.ssg.comcast.net:

Via: SIP/2.0/UDP 76.96.14.7:5060;branch=z9hG4bKmumuql30c0o1ufkph4g0.1

Via: SIP/2.0/UDP SLKDUTQSPS0aa.ssg.comcast.net:5060;received=67.178.70.104;branch=z9hG4bK_1115101141894840168
From: "RTS 1 Port 14"<sip:4352490949@SLKDUTQSPS0aa.ssg.comcast.net;user=phone>;tag=1_1115_f10114_k619_CtkM94DCAE
To: <sip:5036985129@snjscabad01aa.ssg.comcast.net;user=phone>
Call-ID: 1284747884@SLKDUTQSPS0aa.ssg.comcast.net
CSeq: 1

INVITE

Max-Forwards: 69
Supported: 100rel,timer,replaces,unknown
Contact: <sip:4352490949;tgrp=7777;trunkcontext=SLKDUTQSPS0aa.ssg.comcast.net@SLKDUTQSPS0aa.ssg.comcast.net:5060;user=phone;transport=udp>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,REFER,UPDATE

Min-SE: 900
Session-Expires: 1800;refresher=uac

P-Asserted-Identity: "RTS 1 Port 14"<sip:4352490949@SLKDUTQSPS0aa.ssg.comcast.net;user=phone>

User-Agent: BTS10200/900-06.00.03.V00 (SIA)

Content-Length: 435
Content-Type: application/sdp

SAG config on the SBC

atl-sbc12# show running-config session-group XYZ.PRI.SBC

session-group
group-name
description
strategy

app-protocol

XYZ.PRI.SBC
XYZLCR
RoundRobin

SIP

state

enabled

dest

XYZ

trunk-group
stop-sag-recurse
sag-recursion

XYZ:PRI.SBC
401,407
disabled

Advantage

Ability to make a test call over a specific carrier

within an APOP
Using a specific CIC we can target a unique
SBC/Peer
No need to change the LCR table to run a test
call
Validate the resolution of a trouble ticket prior to
route the traffic back to that peer

Thank You