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).
- On the System Configurator main screen, choose .
- Configure the settings according to the following table.
Field
Function
Secure RTP Settings
Note:-
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.
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If any SRTP setting is changed, the change will be applied immediately but if an end-user user interface (such as CDT) is currently open with old settings, the new settings will be applied only after the user interface is restarted.
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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. He makes a consultation call to agent B using SRTP, and completes 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. SRTP is not supported on H.323 trunks.
When the option is selected:
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Outgoing calls: The SIP trunk or SIP phone selects the audio video profile from the non-secure AVP(RTP) or secure SAVP(SRTP) profile.
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Incoming calls:
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If only either AVP(RTP) or SAVP(SRTP) profile is offered, the offered one is used.
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If both AVP(RTP) or SAVP(SRTP) profiles are offered, SAVP(SRTP)is used.
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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.
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To add other codecs, choose one of the drop-down list options G.711A, G.711µ, and G.729.
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To change the priority, choose the codec and click Up or Down buttons to move the codec on the list.
Note:-
Define the same priority order as for global codecs for all system’s components, such as trunks or desk phones. Otherwise audio problems may occur when calls are connected between phones prioritizing different codecs.
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The codec defined for a trunk, route, or destination must be on the list of supported codecs.
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The default value was changed from G.729 to G.711A in SP02. The change affects clean installations only.
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The default value was changed to the list of G.711A and G.711µ in SP07. The change affects clean installations only.
Support for multiple codecs was added in SP04.
Prefer G711 for Playing Prompts
(Prior to SP09 Use G.711µ for Playing Prompts)
If the option is selected (or the former Use G.711µ for Playing Prompts is in use) 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.
Changed in SP09.
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. See Installation Guide for more instructions about Quality of Service definition.
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