Showing posts from 2007

Configure Asterisk to receive incoming SIP calls

If you want people from the outside world to be able to contact you via SIP, there are a few things you need to configure. First, in FreePBX setup, click General Settings on the left hand menu, scroll down and select Yes to Allow Anonymous Inbound SIP Calls . The Asterisk configuration file sip.conf defines the parameters for accepting incoming SIP calls. We need to make some changes to this file to correctly process incoming calls. From the Trixbox Admin web page, click Asterisk , Config Edit , then sip.conf on the left hand side. Modify the contents of this file so it reflects what is shown below. [general] bindport=5060 ; UDP Port to bind to bindaddr= ; ( binds to all) disallow=all allow=ulaw allow=alaw allow=gsm allow=ilbc context=from-sip-external callerid=Unknown tos=0x68 ;------------- Ryan's Mods -------------- externip= ;required behind NAT localnet=  ;required behind NAT fromdomain= canreinv

Understanding DNS SRV records and SIP

What are SRV records? DNS SRV records or service records are a type of DNS entry that specify information on a service available in a domain. They are typically used by clients who want to know the location of a service within a domain. For example, in an Active Directory environment, domain joined windows PCs rely on SRV records to locate domain controllers to authenticate to within their domain. A SRV record record contains the following information: Service Name: the well know name of the service Protocol: specifies if this is a TCP or UDP service Domain Name: the domain name that this record belongs to TTL: Time to Live value Class: DNS class field. This always has the value of "IN" Priority: when multiple hosts are configured for the same service, the priority determines which host is tried first Weight: A relative weight for records with the same priority Port: the TCP or UDP port that the service uses Target: the name of the host providing the service He

Understanding the relationship between SIP and RTP

Getting your head around SIP and RTP traffic flows is a little daunting at first, but its actually not all that complicated when you understand the purpose of the protocols. As its name implies, the Session Initiation Protocol is used to initiate a session between two endpoints. SIP does not carry any voice or video data itself - it merely allows two endpoints to set up connection to transfer that traffic between each other via the Real-time Transport Protocol (RTP). The SIP protocol can be, and usually is, routed through one or more SIP proxy servers before reaching its destination. It is very similar to how email is transmitted, in that multiple email servers are usually involved in the delivery process, each forwarding the message in its original form. Each email server adds a Received header to the message, to track the route the message has taken. SIP uses a Via header to track the SIP proxies that the message has passed through to get to its destination. SIP uses a very si