VoIP Signalling Sample Clauses

VoIP Signalling. SIP over UDP is required. The following standards are adhered to by MTN SA, and is expected of the peering partner to be fully supported: • RFC3261 (V2) Base SIP RFC • RFC2976 INFO Method • RFC3262 PRACK • RFC3323 SIP Privacy • RFC3324 Short Term Requirement for Network Asserted Identity • RFC3325 Trusted Private Networks and SIP • RFC3398 ISUP to SIP mapping • RFC3515 Refer Method Comply • RFC3824 Using E.164 numbers with SIP • RFC3840 Indicating user agent capabilities • RFC2806 URLs for telephone calls • RFC3264 Answer Offer model • RFC3311 UPDATE Method • RFC4028 Session Timer • SIP Options Method Call admission control shall be enforced at the network edge. SIP code “503 Service Unavailable” shall be returned for rejected Calls. The following codecs shall be supported, in order of preference for originated calls: Preference Codec 2 G.729 A Annex B 3 G.729 4 G.729 Annex B 5 G.711 Table 1 List of Codecs Supported For terminated calls, MTN SA will honour whatever is offered in order of the calling party order, as long as the offered Codecs are from the supported list from table 1 VAD or Silence Suppression should as a preference been turned off for any variant of codec if supported by the codec Sampling rates for codecs will be the default for the Codec – i.e. G729 and G711 at 20ms industry standard. No Transcoding or Transrating function is provided between codecs or sampling rates other than mentioned. FAX encoding to be t38. DTMF shall be supported with RFC2833 – Codec identifier 101 Connectivity Connectivity media shall be agreed upon by both parties. (Deviation from the below must be agreed on by both parties.)
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Related to VoIP Signalling

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