The Marratech Manager SIP Module allows SIP users to be included into e-meetings via voice. A SIP user can e.g. be a software or hardware SIP phone called directly or normal phone access via a SIP-PSTN gateway. The gateway can either be provided in-house or via an external SIP-phone gateway provider. Please consult the Marratech support section and the Marratech web forum for more information about compatible SIP-phone gateway providers.

Each call will require a seat just as for normal Marratech users. I.e. if a seats license model is utilized, and no more seats are available then no more calls can be placed. Furthermore, access to the Dial Out capabilities can be controlled on a per user, per user group basis.

Although all communication within a Marratech -meeting is encrypted (optionally defined per e-meeting) it will go unencrypted between the Manager and the SIP Server. This is due to the lack of deployed SIP services that support encryption of control and data traffic.

Meeting rooms defined to use a URI encryption key do not support Dial Out functionality, since the Manager will be unaware of the encryption key required to connect the external SIP users.

SIP users will get all the active audio streams mixed into one audio stream in the PCM audio encoding. Audio originating from a SIP user will show up as a participant of its own in the Marratech interface.