| | Asterisk | OpenSIPS |
| |  |  |
| Architecture | Asterisk functions as a Back to Back User Agent (B2BUA) style architecture is more complex because it is responsible for signaling and the media (voice). In a SIP proxy, such B2BUA services such as as codec translation (ie uLaw to G.729a), protocol translation (SIP to IAX2), and services related to media such as IVR, voicemail, automatic call distribution, text to speech, and voice recognition | OpenSIPS is a Session Initiation Protocol (SIP) Proxy. This makes all the difference between them. The architecture for a SIP proxy is simpler than a B2BUA because a SIP Proxy only deals with signaling |
| Connectivity to the PSTN | Asterisk has many telephony interfaces cards to connect to the PSTN. It's very easy to install telephony cards and configure it in the Asterisk server. | OpenSIPS always need a SIP gateway to connect to the PSTN. There is no possibility to install telephony cards in the server. Often, I will implement Asterisk as a PSTN gateway for OpenSIPS. |
| NAT Traversal | Network Address Translation (NAT) is SIP's Achilles heel. NAT is a function of a network router to map private (local) IP addresses to public (internet) IP addresses. Asterisk can deals with NAT traversal to allows you to send the voice (or video stream) from your IP Telephone or VoIP software client directly to a VoIP telephone service provider. | OpenSIPS deals a lot better with NAT traversal. OpenSIPS allows you to send the voice (or video stream) from your IP Telephone or VoIP software client directly to a VoIP telephone service provider using in most cases (non-symmetric NAT). OpenSIPS also allows you to directly manipulate SIP communication to handle special cases, such as, when you have two VoIP telephones behind the same NAT router and want to send the media directly between them. |
| Load Balancing | Asterisk can be configured to load balance by the "username", "ruri", "callid", and other properties and also failover. | A unique strength of OpenSIPS is its ability to load balance VoIP calls. Specialized hashing algorithms in OpenSIPS can be configured to load balance by the "username", "ruri", "callid", and other properties. OpenSIPS is "failover aware" and can make for a very complementary part of an Asterisk solution. |
| Low Level Access to Header and Transactions | With Asterisk you have low level access to the IAX protocol. This makes it possible to transfer Header signal and Media in same packet and same port! | With OpenSIPS, you have low level access to the Session Initiation Protocol. You can handle all the requests and responses. This makes it possible to translate between two incompatible versions of SIP, handling directly the SIP headers, requests, and responses. This is an important feature when you have SIP implementations from different manufacturers, which can be incompatible between each other. |
| Integration with Radius, Diameter, and LDAP | Asterisk has built-in integration with LDAP, Radius, and Diameter. | OpenSIPS has built-in integration with LDAP, Radius, and Diameter |
| Media Services | Asterisk has capability of any media related (the voice portion of a telephone call) services. | OpenSIPS is a SIP Proxy and is not capable of any media related (the voice portion of a telephone call) services. OpenSIPS is not a platform that you would use to create systems such as IVR, VoiceMail, TTS, and Voice Recognition, etc. This is by design. |
| Conclusion | Asterisk is a Back to Back User Agent (B2BUA) and is very popular in the small to medium PBX market. Asterisk is simpler to configure and can be used as a "black box does it all". | OpenSIPS is a Carrier Class SIP Proxy used primarily by VoIP providers. It is designed to handle large volumes of calls, load balance SIP communication, solve advanced NAT scenarios, and to deal with SIP signalling as no other.OpenSIPS is typically used when you have some special need, such as load balancing or when you are managing large volumes such as more than a thousand registered users. |
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