Introduction into SIP protocol

What Michal knows about SIP?
© 2012 SOLARWINDS WORLDWIDE, LLC. ALL RIGHTS RESERVED.
Brief introduction

SIP from 10 KM
» What is SIP and how it works?
» SIP troubleshooting
» SIP Trunking
2

What is SIP?
» In the world of VoIP, SIP is a call setup protocol that operates at the
application layer.
» Designed to set up real-time multimedia sessions between groups
of participants. Can also be used to set up video and audio
multicast meetings, or instant messaging conferences.
» Calls (voice or video) between SIP users are free, worldwide. There
are no boundaries and no restrictive laws or charges.
» Contrasted with H.323, SIP is an exceedingly simple protocol.
» SIP Applications and SIP addresses are for FREE.
3

How SIP Works?
» You have to have SIP Address and Account (more details later)
» You have to have SIP Client (computer, mobile app, phone)
» You have to configure your SIP Client (via easy config wizard)
» You need a solid bandwidth – at least DSL connection
» You need to tell somebody else your SIP address 
4

SIP Credentials
» Set during SIP provider registration. Must contain “@”.
» User name has to be unique (ideally email).
» Example how SIP credentials may look like:
SIP address, e.g. Michal.Hrncirik@solarwinds.com
Password : e.g. NASYPMI
Username: e.g. Michal.Hrncirik@solarwinds.com
Domain/Realm: e.g. thatsipservice.info
Outbound proxy: e.g. proxy.proksi.com
XCAP root: https://xcap.proksi.com/xcap-root
5

Structure of SIP Address
» SIP:user@domain:port
» “user” – letters or a number
» “@” is mandatory! Splits user and domain.
» “domain” – service you are registering with. Could be name or
IP Address
» “port” – is optional. Defines access on a proxy server.
6

Let’s make a call!
» We are set now, we have SIP Address and configured client!
7

Functions of SIP
User location and registration End points (telephones) notify SIP proxies of their
location; SIP determines which end points will
participate in a call.
User availability SIP is used by end points to determine whether they
will “answer” a call.
User capabilities SIP is used by end points to negotiate media
capabilities, such as agreeing on a mutually supported
voice codec.
Session setup SIP tells the end point that its phone should be
“ringing;” SIP is used to agree on session attributes
used by the calling and called party.
Session management SIP is used to transfer calls, terminate calls, and change
call parameters in mid-session (such as adding a 3-way
conference).
8

SIP Protocol Methods
» REGISTER – registration of caller on a SIP Proxy server
» INVITE – starts communication about scheduled relation
» ACK – start of new relation
» CANCEL – interrupt of initialization before its actual start
» BYE – ends relation
» OPTIONS – ask for attributes of call participants
9

SIP connection flow
10

SIP error codes
» 1xx – initiating, step continuing OK but it hasn’t finished yet
» 2xx – success, step has been finished without issues
» 3xx – redirect, step is in progress but it’s awaiting some data
» 4xx – client error, request failed, server can’t process it
» 5xx – server error, client request OK, server problem
» 6xx – fatal error, completely fatal error can’t be processed.
11

Good to know about SIP
» Purely text based (easy to debug and get error codes!)
» Can run over IPv4 and IPv6
» Can use both TCP or UDP (very likely IPv4 over UDP)
» SIP devices can talk to each other directly (but proxy is 99%)
» SIP proxy deals only with “messaging”. Doesn’t transfer call.
» SIP proxy makes client implementation easier (translation of
extension into SIP address – IP + location and type of device).
» Proxy CAN provide mid-call features: conferencing, accounting
» SIP dos NOT do call control features – voice mail, auto-
attendant. This is feature of end-points.
12

After SIP deployment
Woaa! I made a SIP call and before an hour I didn’t even know I
can do it from my network!
13

Typical SIP problems after deployment
Poor audio quality or user cant’ connect
1) Interoperability – various vendors problem. Basic connection
2) Codec Negotiation – codec support by end-points
3) Slow Error Recovery – UDP effect
4) No Conference control – has to have SIP server/PBX
5) Others:
» Echo on call, Registration is dropping, Timeouts
» Incoming call doesn’t hear you
» Incorrect server settings, Wrong username/password
» DNS and network issues – check your routers accessibility
» Firewall – SIP ALG rules, ISP issues
14

SIP Troubleshooting - connection
» SIP connection and RTP – use Wireshark (use VoIP Calls view)
» Various reasons – format of options, codec issue. But it’s easy
to read.
15

SIP Troubleshooting - quality
» Focus on RTP data (example with wireshark)
» Get the codec used
» Select “RTP-Stream analysis” from Wireshark menu. Check
Jitter, packet loss. Eliminate network issue vs. hardware.
16

SIP Call reconstruction – Wireskark in action
17

SIP Trunking is what companies care about
18

SIP Trunking – what it is?
» Using SIP for delivery Unified Communication services to the
customer (voice, video, desktop sharing, etc.)
» VoIP over “shared” wire
» Save $$$ comparing to PSTN telephony.
» Easy switch – keep your PBX
19

3 Typical problems
» Trunks dropping intermittently is a provider problem. If it happens consistently, check
with the provider first before approaching the IP PBX or SBC vendors.
» One-way audio is most likely a configuration issue, poor data entry, or just plain
negligence. It could be a misconfiguration in the IP PBX, SBC, or at the trunk provider.
» Poor voice quality created by latency, jitter or packet loss may be the result of not
enough trunk bandwidth being allocated.
» These are three less likely issues:
 Codec mismatch is a configuration problem that can occur in the IP PBX, SBC or at the provider’s site.
 Trunk registration failures are a provider support issue.
 No audio is probably another configuration issue. It can be in the IP PBX, SBC or at the provider site.
» The least common issues are:
 ITSP SIP server failures are the provider’s responsibility to prevent.
 Incoming call transfer failure is another provider responsibility to ensure works properly.
 Call conferencing with an external caller failing is another provider responsibility to resolve.
20

SIP Trunking problems (survey 850 WW resp.)
21

Example of failures
» Use 'debug ccsip messages'. (leverage CUBE to do it)
» Session protocol UDP. INVITE from CUCM, No Offer SDP in
INVITE. It means Delayed Offer, in case of Early Offer SDP is
included in INVITE.
» Delayed Offer. Offer SDP in 200 from remote site. Answer SDP
in the following ACK from CUCM
22

SIP Trunk Monitoring
» Identify top failures mentioned above.
» Volume of calls per SIP trunk in a time (spikes).
» Pair the call with the SIP trunk ID – for provider reference.
23
# of Phones
SIP Trunks
Needed
2 4
5 6
10 6
25 12
50 23
100 30

SIP & how to do it?
» Q & A
24

Introduction into SIP protocol

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