When a user puts a call on hold, the SIP ALG releases SDP media resources, such as pinholes and translation contexts. When the user resumes the call, an INVITE request message negotiates a new SDP offer and answer and the SIP ALG reallocates resources for the media stream. This can result in new translated IP address and port numbers for the media description even when the media description is the same as the previous description. This is compliant with RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP).
Some proprietary SIP implementations have designed call flows so that the user agent (UA) module ignores the new SDP INVITE offer and continues to use the SDP offer of the previous negotiation. To accommodate this functionality, you must configure the device to retain SDP media resources when a call is put on hold for reuse when the call is resumed.
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To retain SIP hold resources, use either the J-Web or CLI configuration editor.
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Use the following command to accommodate proprietary SIP call flows: