introduction into sip protocol

© 2012 SOLARWINDS WORLDWIDE, LLC. ALL RIGHTS RESERVED.

What Michal knows about SIP? Brief introduction

© 2012 SOLARWINDS WORLDWIDE, LLC. ALL RIGHTS RESERVED.2

SIP from 10 KM

» What is SIP and how it works?

» SIP troubleshooting

» SIP Trunking


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.


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


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. [email protected] Password : e.g. NASYPMIUsername: e.g. [email protected] Domain/Realm: e.g. thatsipservice.info Outbound proxy: e.g. proxy.proksi.com XCAP root: https://xcap.proksi.com/xcap-root


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.

sip:user@domain:port


Let’s make a call!

» We are set now, we have SIP Address and configured client!


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).


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


SIP connection flow


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.


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.


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!


Typical SIP problems after deployment

Poor audio quality or user cant’ connect1) Interoperability – various vendors problem. Basic connection2) Codec Negotiation – codec support by end-points3) Slow Error Recovery – UDP effect4) No Conference control – has to have SIP server/PBX5) 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


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.


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.


SIP Call reconstruction – Wireskark in action


SIP Trunking is what companies care about


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


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.


SIP Trunking problems (survey 850 WW resp.)


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


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.

# of Phones SIP Trunks Needed

2 45 6

10 625 1250 23

100 30


SIP & how to do it?

» Q & A

introduction into sip protocol

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