Howto:How to use media-relay functionality for sip-registrations and how to configure sip-clients and sip-trunk

From innovaphone-wiki

Jump to: navigation, search

Contents

Summary

This document describes how to configure a sip-user on telephones and a sip-trunk-line in the innovaphone-PBX! It also describes how to configure the media-relay functionality.

Applies To

This information applies to

  • IP 3000, V4
  • IP 400, V4
  • IP 21, V4

Build 01-4001 and later.


More Information

Problem Details

1. Regarding media-relay functionality

a. Due to its special trickiness here is a small guideline on the configuration of SIP trunk interfaces towards the public internet. This concerns "Register as "interfaces to a SIP provider as well as interfaces without registration (e.g. ENUM).

b. Let's start discussion with a quick look at internal-only telephony. Afterwards we're going to add a SIP trunk to our installation.

### Internal IP telephony

Sig Sig
+------------->PBX<------------+
| |
EP <------------------------> EP
RTP


On the internal-only-telephony the RTP streams go always end-to-end between the talking endpoints, not through the PBX in between. The PBX simply forwards the internal RTP address of each endpoint to the other side. This is done for scalability resaons.


### Internal IP telephony with external VOIP trunk

Sig Sig Sig :
+------------->PBX<----------->GATEWAY<---------:----------+
| : |
EP <----------------------------->|<------------:-------> EP
RTP RTP :
:
(private network) : (public network)

c. Internal IP telephony with external VOIP trunk The gateway with the VOIP interface towards the external/public internet must have a public ip-address. Either a real public ip-address on one of its ip interfaces or a mapped public ip-address by means of STUN. In opposite to the internal-only-telephony where the RTP streams go directly between the talking endpoints, the gateway must not give the RTP address of the internal endpoint to the external endpoint, since the external endpoint cannot send to that internal ip-address. The gateway must rather give its own public ip-address to the external endpoint. The gateway then receives RTP from the external endpoint on its public ip address and forwards it to the internal endpoint. This is what we call "media-relay".

In order to use the media-relay functionality, the gateway must be able to recognize sessions between internal and external endpoints. To achieve that the gateway must know the address range of the internal network. System Requirements

V6 Build on gateways and telephones is absolutely needed.

An existing sip-account is needed. For example a user with user-id (URI) 8111111e0@sipgate.de. and password 1234. Configuration

1. Configuring the media-relay functionality

a. The administrator must configure this ip-address range on: IP/Settings/Private Networks. In addition to the real internal network addresses, the internal loopback address range must be configured there. Example: 192.168.0.0 / 255.255.255.0 127.0.0.0 / 255.255.255.0

2. Configuring a sip-user to an innovaphone VoIP-Telephone

a. Go to the telephone web-interface and switch to Configuration/Registrationx/Registration tab and enable it.

b. As Protocol select SIP

c. As primary ip-address you can enter the sip-provider ip-address there if want to send the REGISTER request to another than configured in domain. This is not needed if a domain is configured because the ip-address could be resolved from the entered domain (DNS). If you want to use the domain only for the SIP-URI creation and send the REGISTER to an specific ip-address enter an ip-address and it will override the domain, without doing any DNS resolving. If your sip-provider supports a secondary ip-address you can configure an alternative ip-address if the primary ip-address isnt available.

d. Defines the local ip-address, the telephone uses for registration. This setting is only for exceptional cases.

e. As domain enter the sip-provider web-address (e.g. sipgate.de) which you can find in the uri-string (term behind @) given by the sip-provider.

f. As user-id enter the number (e.g. 8111111e0) which you can find in the uri-string (term before @) given by the sip-provider (e.g. 1234).

g. As stun server enter the stun-server web-address given by the sip-provider.

h. For correct authorisation enter the user-name (user-id) and password into the Authorization fieldset.

i. After clicking OK the state should be up displayed in the right upper corner. Notice a sip-call dont support dial-tones. You need to enter the wished number before dialling.

3. Configuring a sip-trunk-line to an innovaphone VoIP-Gateway

a. First configure a pbx-object trunk e.g. with h.323-name SIPTrunk and E.164-number 8.

b. Then go to Administration/Gateway/Interfaces and select on of the SIPx-interfaces.

c. Enter a descriptive name.

d. As primary ip-address you can enter the sip-provider ip-address there if want to send the REGISTER request to another than configured in domain. This is not needed if a domain is configured because the ip-address could be resolved from the entered domain (DNS). If you want to use the domain only for the SIP-URI creation and send the REGISTER to an specific ip-address enter an ip-address and it will override the domain, without doing any DNS resolving. If your sip-provider supports a secondary ip-address you can configure an alternative ip-address if the primary ip-address isnt available.

e. As ID configure the number (e.g. 8111111e0) which you can find in the uri-string (term before @) given by the sip-provider. Behind the @-symbol enter the sip-provider web-address (e.g. sipgate.de) which you can find in the uri-string (term behind @) given by the sip-provider.

f. As account enter the user-id (e.g. 8111111e0) which you can find in the uri-string (term before @) given by the sip-provider.

g. As stun server enter the stun-server web-address given by the sip-provider.

h. If Account is not set, this Name (part in the uri-string before the @-symbol) is used for the registration.

i. As Account enter the user-id (e.g. 8111111e0) which you can find in the uri-string (term before @) given by the sip-provider.

j. For correct authorisation enter the password (e.g. 1234) of the sip-account in the password field and retype it.

k. Now it is needed to bind this sip-registration to a pbx-sip-trunk-line. To get this work enable the h.323-registration and configure the primary and if existing the secondary gatekeeper ip-address (e.g. 127.0.0.1 if the pbx is running on the same device). It is adequate to configure only the gatekeeper-id.

l. At least the information of the h.323-name or e.164-number is needed. m. After clicking OK the state should display the ip-address of the sip-provider. As alias it should display the name and number of the pbx-trunk-object (e.g. SIPTrunk:8). Under registration the ip-address of the pbx (e.g. 127.0.0.1) is displayed. In the pbx-object-overview the trunk-line-object should also be registered. Notice a sip-call dont support dial-tones. You need to enter the wished number before dialling.


Known Problems

1. Regarding media-relay functionality Working with giving private ip-addresses to the outside world could work under some very special circumstances, but is not recommended! You probably will have strange problems like "one way media" or "dying registrations" or "no incoming calls".

2. Regarding IP202 The ip202 doesnt have a pbx component anymore. So it is only possible to register this phone to a sip-account. It is not possible to use this phone as sip-server (registering sip-telephones to this phone) or to configure it as sip-trunk-line.

Related Articles

Personal tools