Managing RTP and Codec Settings

Use Real-time Transport Protocol (RTP) settings to define RTP-related settings such as codec, and to start using secure RTP (SRTP).

Note: You may not be able to update all of these settings. If you need any changes, contact Sinch.
  1. On the System Configurator main screen, choose Call Switching > Global Switching Settings > Real-time Transport Protocol (RTP).
  2. Configure the settings according to the following table.



    Secure RTP Settings

    • If all components are not able to use SRTP, using SRTP requires an MRS module where the option Use for Server-Side Recording is selected. This is shown as value 1 in the Module view (System Management > Modules, MRS module).

    • If any SRTP setting is changed, the change will be applied immediately but if an end-user user interface (such as CP) is currently open with old settings, the new settings will be applied only after the user interface is restarted.

    • If you use SRTP, make sure that the SIP trunks connected to the contact center system are configured to use only SRTP (such as VOIP Gateway, SBC, operator SIP trunks). Call transfers between RTP and SRTP are not supported. For example: A call from PSTN without SRTP is connected to agent A. They make a consultation call to agent B using SRTP, and complete the call transfer of the PSTN call to the agent B. In this case the audio is corrupted.

    Enable SRTP

    Enables using secure RTP. The only supported cipher suite in the system is the AES-CM-128-HMAC-SHA1-32.

    When the option is selected:

    • Outgoing calls: The SIP trunk or SIP phone selects the audio video profile from the non-secure AVP(RTP) or secure SAVP(SRTP) profile.

    • Incoming calls:

      • If only either AVP(RTP) or SAVP(SRTP) profile is offered, the offered one is used.

      • If both AVP(RTP) or SAVP(SRTP) profiles are offered, SAVP(SRTP)is used.

    Use SRTP after NAT

    Define if SRTP is used for certain NAT locations. This enables using SRTP also between desk phones and trunk.

    Use SRTP between MRS Modules

    Define if SRTP is also used between MRS servers. This enables that dialing between different NAT locations can be secured as well.

    Supported Codecs

    Define the list of global audio codecs used in the system for general coding-decoding analog-digital transformations of voice. This setting applies to all internal and external calls including playing prompts and call recordings. When more than one codec is defined, the system can accept incoming calls with any of the supported codecs, and also offer all supported codecs in outgoing calls. If more than one codec is listed, they are used in prioritized order.

    • To add other codecs, choose one of the drop-down list options G.711A, G.711µ, and G.729.

      Note: G.729 is not supported in Sinch Contact Pro cloud.
    • To change the priority, choose the codec and click Up or Down buttons to move the codec on the list.

    • Define the same priority order as for global codecs for all system’s components, such as trunks. Otherwise audio problems may occur when calls are connected between phones prioritizing different codecs.

    • The codec defined for a trunk, route, or destination must be found on this list of supported codecs.

    Prefer G.711 for Playing Prompts

    If the option is selected, the first one of G.711 or G.711µ found on the list of supported codecs is used for playing prompts by MRS. By default, MRS modules use the defined priority order.

    Other settings

    Prefer Direct RTP

    Prefer direct RTP streams between end-points whenever possible.

    DSCP Value

    Define the system-wide Differentiated Services Code Point (DSCP) value that is used for tagging RTP packets for classification, and enables using Quality of Service (QoS). Enter a value from 0 to 63. The default value is 46.