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




CSeq: 1 INVITE

Max-Forwards: 70


Contact: <sip:4352490949;tgrp=7777;trunk-context=SLKDUTQSPS0aa.ssg.comcast.net@SLKDUTQSPS0aa.ssg.comcast.net:5060;user=phone>


Min-SE: 900




Content-Length: 435


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




CSeq: 1 INVITE

Max-Forwards: 69


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

context=SLKDUTQSPS0aa.ssg.comcast.net@SLKDUTQSPS0aa.ssg.comcast.net:5060;user=phone;transport=udp>


Min-SE: 900




Content-Length: 435


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

2. dreas barache sipnoc2015 keynote v4

Documents

sip routing

sip clients

clients sip

webrtc sip

sip trunking

sip peers commercial

network servers

deployed sip pbx