2. Dreas Barache SIPNOC2015 Keynote v4

June 24th, 2015

Comcast SIP Operations

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

2

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 3

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

4

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

5

Sample Dashboard

6

What’s 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!!!

7

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

Call Flow ENUM Query

SB

C

SRP CMS/IMS Peer

SBC

IRDB

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;[email protected]: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:[email protected];user=phone>;tag=1_1115_f10114_k619_CtkM94DCAE

To: <sip:[email protected];user=phone>

Call-ID: [email protected]

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:[email protected];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;[email protected]: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:[email protected];user=phone>;tag=1_1115_f10114_k619_CtkM94DCAE

To: <sip:[email protected];user=phone>

Call-ID: [email protected]

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:[email protected];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;[email protected] ttl= 60

SNJ-SPRXY-02#

Target SAG within the SRP

group-name WDSTGAHQ-INT.SBC.SIP.SAG

description Peer-XYZ Woodstock 9200 SBC SAG

state enabled

app-protocol SIP

strategy RoundRobin

dest

WDSTGAEMBSCaa.ssg.comcast.net

WDSTGAEMBSEaa.ssg.comcast.net

WDSTGAEMBSKaa.ssg.comcast.net

trunk-group

sag-recursion disabled

stop-sag-recurse 401,407

INVITE sent to the SBC INVITE sip:5036985129;tgrp=XYZ;trunk-context=PRI.SBC;npdi;rn=5036583783;[email protected]:

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:[email protected];user=phone>;tag=1_1115_f10114_k619_CtkM94DCAE

To: <sip:[email protected];user=phone>

Call-ID: [email protected]

CSeq: 1 INVITE

Max-Forwards: 69

Supported: 100rel,timer,replaces,unknown

Contact: <sip:4352490949;tgrp=7777;trunk-

context=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:[email protected];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 XYZ.PRI.SBC

description XYZLCR

strategy RoundRobin

app-protocol SIP

state enabled

dest XYZ

trunk-group XYZ:PRI.SBC

stop-sag-recurse 401,407

sag-recursion 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