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Connecting Your Enterprise With Asterisk: IAX to Carriers - Asterisk-ES

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<strong>Connecting</strong> <strong>Your</strong> <strong>Enterprise</strong><br />

<strong>With</strong> <strong>Asterisk</strong>: <strong>IAX</strong> <strong>to</strong> <strong>Carriers</strong><br />

Day<strong>to</strong>n Turner<br />

Voxter Communications


What is <strong>IAX</strong><br />

Inter <strong>Asterisk</strong> eXchange<br />

Developed by Digium<br />

and the Open Source<br />

Community<br />

Alternative <strong>to</strong> SIP,<br />

H.323<br />

Pronounced “eeks”


Where is <strong>IAX</strong> used<br />

Between <strong>Asterisk</strong><br />

Servers for inter-PBX<br />

communication<br />

Links <strong>to</strong> your ITSP<br />

<strong>IAX</strong>y - Digium’s <strong>IAX</strong><br />

enabled ATA<br />

Soft Phones, some<br />

hard phones


Who Implements <strong>IAX</strong><br />

<strong>Asterisk</strong> (of course)<br />

FreeSWITCH<br />

Yate<br />

SofaSwitch<br />

OPAL<br />

No commercial vendors (yet!)


Benefits of <strong>IAX</strong><br />

Single Port (UDP 4569), makes for easy scalability!<br />

Advanced Media Transfers<br />

“Real” trunking!<br />

Encryption (A<strong>ES</strong>128)<br />

Authentication (Plaintext, MD5, RSA)


Scalability<br />

Load Balance-able<br />

(iax-proxy, LVS, etc)<br />

Dynamically Sized<br />

Thread Pool<br />

Binary Encoded for<br />

efficiency


Comparison: SIP vs <strong>IAX</strong><br />

Bandwidth Usage<br />

Codec<br />

SIP<br />

<strong>IAX</strong> (Trunked)<br />

1st Call Additional Calls 1st Call Additional Calls<br />

G.711 (64kbps) 80kbps 80kbps 80kbps 64kbps<br />

G.726 (32kbps) 48kbps 48kbps 46kbps 32kbps<br />

G.729 (8kbps) 24kbps 24kbps 23kbps 8kbps<br />

G.722 (64kbps) 80kbps 80kbps 80kbps 64kbps<br />

GSM (13kbps) 29kbps 29kbps 28kbps 13kbps<br />

* Bandwidth includes IP overhead, and accounts for only one side of the call. Total usage is double the shown<br />

value since VoIP traffic usage is symmetric.


Comparison: SIP vs <strong>IAX</strong><br />

Bandwidth Usage, Total Calls (G729)<br />

240<br />

SIP<br />

<strong>IAX</strong> (Trunked)<br />

240<br />

180<br />

120<br />

93<br />

120<br />

84<br />

60<br />

30<br />

13<br />

5 10<br />

42<br />

32<br />

0<br />

128kbps 256kbps 768kbps 1mbit 2mbit


Comparison: SIP vs <strong>IAX</strong><br />

Bandwidth Usage, Total Calls<br />

Codec<br />

SIP<br />

<strong>IAX</strong> (Trunked)<br />

DSL T1 DSL T1<br />

G.711 (64kbps) 9 19 11 23<br />

G.726 (32kbps) 16 32 23 47<br />

G.729 (8kbps) 32 64 93 190<br />

G.722 (64kbps) 9 19 11 23<br />

GSM (13kbps)<br />

26 52 57 117<br />

* DSL bandwidth presuming 768kbps available, T1 presuming 1.5mbps


<strong>IAX</strong> Pro’s<br />

Bandwidth: <strong>IAX</strong> Trunks, SIP does not.<br />

Network Configuration: <strong>IAX</strong> traverses NAT and firewalls<br />

with ease. SIP requires more effort (STUN, ICE, TURN)<br />

Internationalization: <strong>IAX</strong> sends language info in headers<br />

QoS: <strong>IAX</strong> gathers its own performance stats (latency,<br />

jitter measurements)<br />

Remote Dialplan: <strong>IAX</strong> can ask a peer about its dial plan,<br />

allowing dialplans <strong>to</strong> be centralized


SIP Pro’s<br />

SIP has been around longer and has much greater<br />

adoption in the industry<br />

Greater numbers of hardware manufacturers (PBX, IP<br />

Phones) implement SIP than <strong>IAX</strong> in their equipment<br />

There is a much more broad audience looking at and<br />

using SIP. Because of this you will find many more SIP<br />

<strong>to</strong>ols (diagnostic, moni<strong>to</strong>ring, load testing, etc) than <strong>IAX</strong><br />

<strong>to</strong>ols.


Planning your <strong>IAX</strong> setup<br />

Codec Selection<br />

Audio Quality or Bandwidth Efficiency<br />

CPU - Are we going <strong>to</strong> transcode<br />

QoS<br />

LAN<br />

Switches that honor QoS (DiffServ), set ToS bits in <strong>Asterisk</strong><br />

WAN<br />

Traffic shaping at your router, consider your endpoints.


Topology Example<br />

Voice<br />

Gateway<br />

(<strong>Asterisk</strong>)


Topology Example<br />

Voice<br />

Gateway<br />

(<strong>Asterisk</strong>)<br />

Internet<br />

PSTN<br />

(T1 PRI)


Topology Example<br />

Client<br />

(ADSL)<br />

Voice<br />

Gateway<br />

(<strong>Asterisk</strong>)<br />

Internet<br />

PSTN<br />

(T1 PRI)


Topology Example<br />

Client<br />

(ADSL)<br />

Voice<br />

Gateway<br />

(<strong>Asterisk</strong>)<br />

Internet<br />

PSTN<br />

(T1 PRI)


Topology Example<br />

Client<br />

(ADSL)<br />

Voice<br />

Gateway<br />

(<strong>Asterisk</strong>)<br />

Peering<br />

Internet<br />

PSTN<br />

(T1 PRI)


Topology Example<br />

Client<br />

(ADSL)<br />

Voice<br />

Gateway<br />

(<strong>Asterisk</strong>)<br />

Peering<br />

DSL<br />

Provider<br />

Internet<br />

PSTN<br />

(T1 PRI)


Topology Example<br />

Client<br />

(ADSL)<br />

Voice<br />

Gateway<br />

(<strong>Asterisk</strong>)<br />

Peering<br />

DSL<br />

Provider<br />

Internet<br />

PSTN<br />

(T1 PRI)


Topology Example<br />

Client<br />

(ADSL)<br />

Voice<br />

Gateway<br />

(<strong>Asterisk</strong>)<br />

Peering<br />

DSL<br />

Provider<br />

SIP<br />

Provider<br />

Internet<br />

PSTN<br />

(T1 PRI)


Topology Example<br />

Client<br />

(Far Away)<br />

Voice<br />

Gateway<br />

(<strong>Asterisk</strong>)<br />

Peering<br />

DSL<br />

Provider<br />

SIP<br />

Provider<br />

MPLS<br />

Internet<br />

PSTN<br />

(T1 PRI)


Config Example<br />

Client<br />

Server<br />

register => clientname:mysecret@myitsp.com<br />

[servername]<br />

type=friend<br />

host=myitsp.com<br />

secret=mysecret<br />

notransfer=yes<br />

dtmfmode=rfc2833<br />

context=inbound<br />

qualify=yes<br />

trunk=yes<br />

disallow=all<br />

allow=g729<br />

[clientname]<br />

type=friend<br />

host=dynamic<br />

secret=mysecret<br />

notransfer=yes<br />

dtmfmode=rfc2833<br />

context=outbound<br />

qualify=yes<br />

trunk=yes<br />

disallow=all<br />

allow=g729


<strong>IAX</strong> Capable ITSPs<br />

Voxter Communications - POPs in Vancouver, BC,<br />

Canada, Seattle WA, Phoenix AZ, full North American<br />

Termination/Origination<br />

VoicePulse<br />

Tel<strong>IAX</strong><br />

More listed at voip-info.org


Thanks for coming!<br />

Any questions

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