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Technical Guide Voice Inbound Service Version 1.1 Sophisticated Simplicity In Mobile Interaction Sophisticated Simplicity In Mobile Interaction

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Page 1: Technical Guide Voice Inbound Service - Tyntec · 5. Audio formats tyntec provides a wide range of audio formats with which the audio data can be transported. The format used is negotiated

Technical GuideVoice Inbound Service

Version 1.1

Sophisticated Simplicity In Mobile InteractionSophisticated Simplicity In Mobile Interaction

Page 2: Technical Guide Voice Inbound Service - Tyntec · 5. Audio formats tyntec provides a wide range of audio formats with which the audio data can be transported. The format used is negotiated

Table of Contents

1. Introduction to Voice Inbound Service 3

2. Requirements for Voice Inbound 3

3. Technical Overview 43.1. General Interface Information3.2. tyntec specific Interface Information

4. Detailed Communication Process 5

5. Audio formats 5

7. SDP 6

8. Telephone Numbering Plan 6

9. Health check 7

10. Load Balancing 8

11. Frequently Asked Questions 8

12. Glossary of Terms 9

13. Contact Information 9

Page

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1. Introduction to Voice Inbound Service

tyntec enables companies to integrate voice services into the core of what they do by acting as a gateway between PSTN and VoIP using SIP, SDP and RTP.

All access is realized via IP, without the hassle of worrying about the legacy telephony protocols. tyntec receives voice calls on your behalf and converts the TDM calls to Voice over IP according to the SIP standard (SIP RFC3261, SDP RFC2327, RTP RFC3550).

This document describes the interface depicted with a blue arrow in the picture.

2. Requirements for Voice Inbound

To implement and use the Voice Inbound service, the customer should have certain resources available that are not provided by tyntec. Principally, they require a SIP user agent implementation, of which there are many commercial and open source solutions on the market.

Additionally, tyntec needs to know the static IP address of the customer‘s server. Connections to FQDNs or host names are not possible. This is because during the setup process tyntec white-lists the IP addresses in the firewall. If the customer can provide multiple IP addresses tyntec is able to balance the incoming calls over the servers and to do fail-over to other IP addresses when the service at one of the IP addresses is not available any more. Please contact customer relations to receive more information about this feature.

Customers should open the necessary IP addresses and ports found in the BIO, because tyntec may send SIP and RTP from different IP ranges.

Dail Virtual Long Number from any device Customer Reach subscriber

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3. Technical Overview

3.1. General Interface Information

SIP is an industry standard defined by the IETF and enjoys wide-spread industry use.

3.2. tyntec specific Interface Information

Acting as a gateway, tyntec transports SIP and RTP data.

The following SIP methods are supported:

Other requests such as, but not limited to, REGISTER, UPDATE are not supported.

The typical call flow between tyntec and the customer is shown in the standard call diagram below. This diagram can be seen as a standard scenario which will apply to most call flows on the platform, but more complex scenarios, like 183 Session progress answers are possible as well.

tyntec uses port 5060 for all SIP messages. For RTP traffic, tyntec uses a port range from 35000 to 65000. SIP and RTP use Datagram (UDP) as the transport layer protocol.

All addresses must be public IP addresses. Hostname resolving and fully qualified domain names (FQDNs) are not supported. In case the customer is connected to tyntec over a VPN, the IP addresses are not public.

Request

INVITE

ACK

BYE

CANCEL

INFO

OPTIONS

Comment

Indicates a client is being invited to participate in a call session.tyntec supports privacy features like CLIR, according to RFC 3325. In case of a forwarded call the diversion key will be filled with a value.

Confirms that the client has received a final response to an INVITE request.

Terminates a call and can be sent by either the caller or the callee.

Cancels any pending request.

Sends mid-session information that does not modify the session state. Can be used for DTMF information.

Queries the capabilities of servers. Request available only for load balancing feature.

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4. Detailed Communication Process

Add a concrete SIP call flow including the SDP portions

Invite

100 Trying

180 Ringing

200 OK

ACK

RTP data

BYE

200 OK

A common scenario for the voice inbound service is the flow of a call on the telephony network that is directed to the customer. In this default case an INVITE request will be send by tyntec‘s platform to the IP(s) provided by the customer. This INVITE request contains all necessary information given by the caller. Information includes, but is not limited to, caller ID, callee ID, privacy level or defer information.

This INVITE should be answered by the customer immediately, the most common case is a 100 Trying response. Thus informing tyntec that customer service has received the INVITE. Often the customer replies with a 180 Ringing response that tells tyntec to play a ring back tone for the caller. Another common reply is 200 OK which informs tyntec‘s platform that customer service is ready to receive a call.

After that tyntec‘s platform sends an ACK request to the customer, which initiates the media data that is transmitted via RTP over UDP.

If the callee ends the call, a BYE request is sent from the customer‘s service to tyntec. This will halt the media transport and it will be answered with a 200 OK response.

Please keep in mind that this is only a sample call flow and details can differ.

5. Audio formats

tyntec provides a wide range of audio formats with which the audio data can be transported. The format used is negotiated between the parties at call set up. Please be aware, that not all audio formats that are theoretically available are activated. If you want a specific audio format, that is not offered by the tyntec platform, please talk to customer relations so that we can discuss adding the additional audio format.Furthermore, it is important to note that there are regional and technical limitations, so not every codec is available for every country or network.A recommended and commonly used audio format is G.711. Upon request, tyntec offers other formats as well, for example G.726, G.729ab, G.723 or iLBC.

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6. DTMF

tyntec supports three different methods of transporting DTMF tones:

as SIP INFO requests (described in RFC 2976) as telephone-event packets in the RTP stream (described in RFC 2833) in-band as part of the audio stream (RTP)

The in-band transmission method is the least proposed, because it sensitive to audio compression and can only be used reliably in combination with G.711 (PCMA/PCMU). Please note that the list of supported methods may differ from country to country, when in doubt please consult with customer support.

7. SDP

tyntec‘s description complies with RFC 3555 and RFC 4566.

Example:Content-Type: application/sdp

v=0o=- 1321474370742 1321474370742 IN IP4 1.2.3.4s=-c=IN IP4 1.2.3.4t=0 0m=audio 41730 RTP/AVP 0 8 18a=nortpproxy:yes

8. Telephone Numbering Plan

The service uses the E.164 numbering plan. A number is made up of the country code (CC), the nation destination code (NDC) and the subscriber number (SN). Numbers have a ‚+‘-sign prefix. The calling party‘s number is found in the From: header field, the called party‘s number in the To: field.

Value

Caller ID (A number)

Called party (B number)

Format

+<CCC><NDC><SN>

+<CC><NDC><SN>

Example

+49231123456

+49231123456

SIP Header

From:

To:

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9. Health check

In order to ensure that there is always a viable connection in place, tyntec performs systematic checks. This is done by sending SIP OPTIONS requests to the customer‘s equipment.

As a default configuration tyntec expects a reply back with a 200 OK response code to the SIP OPTIONS requests. In addition, other response codes can be configured upon the customer‘s request.

SIP request sent by tyntecOPTIONS sip:198.51.100.25 SIP/2.0Via: SIP/2.0/UDP 178.23.25.151:2135;branch=z9hG4bK123456Max-Forwards: 1To: <sip:198.51.100.25>From: <sip:[email protected]>;tag=ABCDE12345Call-ID: 123456CSeq: 1 OPTIONSAccept: application/sdpContent-Length: 0

SIP response expected from customerSIP/2.0 200 OKVia: SIP/2.0/UDP 178.23.25.151:2135;branch=z9hG4bK123456To: <sip:198.51.100.25>;tag=123456From: <sip:[email protected]>;tag=ABCDE12345Call-ID: 123456CSeq: 1 OPTIONSContent-Length: 0

Important note:tyntec uses dedicated IP addresses for the health check. These IP addresses need to be enabled in your firewall configuration in addition to the IP address that are used for normal SIP traffic. For the specific IP addresses please refer to the BIO form.

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10. Load Balancing

As an optional feature tyntec offers to distribute the SIP traffic to different servers of the customer. This way the customer can use many servers to process the incoming calls. tyntec checks the availability of the customer‘s servers periodically with SIP OPTIONS requests. If no answer is received to a series of these health checks the server is marked as unavailable and no new calls will be forwarded to it. The server‘s availability is continuously checked. Once tyntec‘s OPTIONS requests are answered the server is considered to be available (again) and calls will be forwarded to it.

11. Frequently Asked Questions

Q: I do not receive any SIP messages from tyntec despite calling my numberA: You should:Check in the BIO document you got from tyntec if it lists the correct IP addressesCheck if your firewall allows the tyntec IP addresses

Q: No media is transported. Voice clients can‘t hear each other.A: You should:Check whether RTP packets via UDP reach your system, check if they are allowed in the firewall.Check in the SIP messages if the correct IP addresses are stated in the media.

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12. Glossary of Terms

• DTMF: Dual-tone multi-frequency signaling• IETF: The Internet Engineering Task Force http://www.ietf.org/, standard body• MSISDN: Mobile Subscriber ISDN Number.• RFC: Request For Comments – standard document from IETF• RTP: Real-Time Transport Protocol, RFC 3550• SIP: Session Initiation Protocol, RFC3261• SDP: Session Description Protocol, RFC 2327• PSTN: Public switched telephone network standardized by the ITU-T• VOIP: voice over IP - internet telephony using standards such as SIP, RTP, etc.

13. Contact Information

Please contact tyntec’s Customer Relations for the enabling of features and for commercial conditions:

• Email: [email protected] • Phone: +49 700 TYNTECBIZ or +49 89 202 451 200• Fax: +49 89 202 451 101

Page 10: Technical Guide Voice Inbound Service - Tyntec · 5. Audio formats tyntec provides a wide range of audio formats with which the audio data can be transported. The format used is negotiated

Munich +49 (89) 202 451 100 – London +44 (207) 436 0283 – Singapore +65 6478 3020E-Mail [email protected] www.tyntec.com

Sophisticated Simplicity In Mobile Interaction