[asterisk-dev] [Code Review] Asterisk Support of SIP Connect 1.1

[Neeharika Allanki]

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https://reviewboard.asterisk.org/r/1515/
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Review request for Asterisk Developers.


Summary
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This Chan-SIP patch brings Asterisk into compliance with the SIPconnect1.1.  SIPconnect1.1 is a newly released SIP Forum specification that describes a common set of signaling and media interworking procedures for the SIP Trunk interface between a SIP-based IP-PBX and a SIP-enabled Service Provider network. This patch, coupled with specific Asterisk configuration settings, will enable Asterisk to comply with the normative SIP-PBX requirements specified in SIPconnect1.1.

The patch diff listings being submitted are against Asterisk version 1.8.11.The patch itself has been tested against the 1.8.0 version of Asterisk for the following SIPconnect1.1 functions/capabilities:

Security
-TLS
-SIP Digest

Registration (RFC 6140)
-Basic GIN registration
-did not test the GIN interactions with the GRUU and reg-event package extensions)

Calling features
-Basic DID/DOD calls
-Calling name/number delivery with and without privacy
-Early media
-Call Forwarding
-Call Transfer (attended and blind)
-Emergency calls
-DTMF relay


This addresses bug ASTERISK-18705.
    https://issues.asterisk.org/jira/browse/ASTERISK-18705


Diffs
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  /trunk/channels/chan_sip.c 333472 
  /trunk/channels/sip/include/sip.h 333472 
  /trunk/configs/sip.conf.sample 333472 
  /trunk/include/asterisk/strings.h 333472 

Diff: https://reviewboard.asterisk.org/r/1515/diff


Testing
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Thanks,

Neeharika

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