Direct calls between two SIP phones without passing through a SIP proxy
Emin Gabrielyan
2007-04-05
Switzernet Sàrl
Here we show how to configure two Budge Tone-100 SIP phones in order to make direct calls between these two phones. This exercise can be useful for testing VOIP PSTN gateways, such as Cisco routers.
Direct
calls between two SIP phones without passing through a SIP proxy
1. Components of the
current test configuration:
2. How to permit direct
calls:
3. Configuration of the SIP
phone located at 192.168.1.10
4. Configuration of the SIP
phone located at 192.168.1.11
6. Other sections of the
simple SIP tutorial
1. Components of the current test configuration:
We use two Grandstream Budge Tone-100 SIP phones. The phones have static IP addresses: 192.168.1.10 and 192.168.1.11
2. How to permit direct calls:
In order to make outgoing calls without a Proxy server we must permit in the calling phone the outgoing calls without registration. In order to send direct calls to a SIP phone, the calling phone should consider the called phone as its SIP server. In the mean time, the called phone can use a true SIP server. If the called phone is using a true SIP server, it must accept incoming SIP messages from IP addresses other than its SIP server.
Permitting outgoing calls without registration and allowing incoming messages from any IP address are options which can be configured in Budge Tone-100 SIP phones.
The presented example is limited by point to point configuration.
3. Configuration of the SIP phone located at 192.168.1.10
The HTML configuration and status pages of this SIP phone are the following [status], [basic settings], [advanced settings].
The SIP phone is configured with a static IP address:
[htm]
It uses as its SIP server the second SIP phone located at 192.168.1.11.
[htm]
In order to be able to send calls without registration we must “Allow outgoing calls without Registration”. Since there is no true SIP server at 192.168.1.11, our phone will be never registered. Without this option our phone will not give a dial-tone for dialling a phone number (normally the dial-tone appears only after registration with a valid SIP server).
[htm]
4. Configuration of the SIP phone located at 192.168.1.11
This IP phone is configured similarly as the first one. The HTML configuration and status pages of this SIP phone are the following [status], [basic settings], [advanced settings].
The SIP phone is configured with a static IP address:
[htm]
It uses as its SIP server the first SIP phone located at 192.168.1.10.
[htm]
Since there is no true SIP server at 192.168.1.10, this phone will never get registered. In order to be able to send calls without registration we must “Allow outgoing calls without Registration”. In case this phone is registered with a true SIP server, located for example under IP address 192.168.1.15, we can still accept direct inbound calls from the first SIP phone by disabling the following option “Allow incoming SIP messages from SIP proxy only”.
[htm]
5. Conclusions
With the presented configurations, we can make direct calls between the two phones establishing voice communication. See the RTP and SIP message statistics of these two phones [192.168.1.10], [192.168.1.11].
Such configuration must allow you to send test calls directly to a PSTN gateway (such as to a Cisco router) assuming that the Firewall settings of the gateway will pass the calls originated from your IP address.
6. Other sections of the simple SIP tutorial
Examining the STUN settings of a SIP phone
Creating and sending INVITE and CANCEL SIP text messages
* * *
This document [doc], [htm], [ms-htm]