osmo-sip-connector.git branch master updated. dac13bed500ad218143bab3fad95ec2ca316eabd

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gitosis at osmocom.org gitosis at osmocom.org
Sat Mar 26 20:10:29 UTC 2016


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the project "MNCC <-> SIP bridge".

The branch, master has been updated
       via  dac13bed500ad218143bab3fad95ec2ca316eabd (commit)
       via  7a30c94030babb10af3d8293d64011df7a2023a0 (commit)
       via  5f29be5acfa130f704a81da01b6ac5f2b9871458 (commit)
       via  392313ff735af6b17672787c83fa80fe5439710f (commit)
       via  b723cceee964ba57c23be30c7f736dece04b2ce3 (commit)
       via  586abf9f0fba385de1a8b22376a870a556909d97 (commit)
       via  6f039c158deb36a7995c828460bad49f0b9db642 (commit)
       via  b927a0bbdebab81c108059ad6d6a0a812fff2b75 (commit)
      from  989bef0875f3123681bd0d6b1fd8d7ce16fa9eec (commit)

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- Log -----------------------------------------------------------------
http://cgit.osmocom.org/osmo-sip-connector/commit/?id=dac13bed500ad218143bab3fad95ec2ca316eabd

commit dac13bed500ad218143bab3fad95ec2ca316eabd
Author: Holger Hans Peter Freyther <holger at moiji-mobile.com>
Date:   Sat Mar 26 21:09:07 2016 +0100

    sip/app: Route call from SIP to MNCC and deal with the release
    
    Fix releasing of the leg in case it is not routable and make the
    differentation if we initiated the invite (send CANCEL) or send
    a final error. The error code was randomly picked and once we have
    an enum of causes we can decide where to map it to.

http://cgit.osmocom.org/osmo-sip-connector/commit/?id=7a30c94030babb10af3d8293d64011df7a2023a0

commit 7a30c94030babb10af3d8293d64011df7a2023a0
Author: Holger Hans Peter Freyther <holger at moiji-mobile.com>
Date:   Sat Mar 26 20:55:02 2016 +0100

    sip: Continue SIP->MNCC (MO or SIP Originated) implementation
    
    Copy out to/from numbers, copy the sdp file, set the magic for the
    nh handle. With the incoming cancel the leg and the entire call will
    be terminated.

http://cgit.osmocom.org/osmo-sip-connector/commit/?id=5f29be5acfa130f704a81da01b6ac5f2b9871458

commit 5f29be5acfa130f704a81da01b6ac5f2b9871458
Author: Holger Hans Peter Freyther <holger at moiji-mobile.com>
Date:   Sat Mar 26 20:30:14 2016 +0100

    sip: Actually return after having rejected the new call

http://cgit.osmocom.org/osmo-sip-connector/commit/?id=392313ff735af6b17672787c83fa80fe5439710f

commit 392313ff735af6b17672787c83fa80fe5439710f
Author: Holger Hans Peter Freyther <holger at moiji-mobile.com>
Date:   Sat Mar 26 20:02:06 2016 +0100

    sdp: Extract SDP traversing into a separate module
    
    The code will still know about SIP and the sip_call_leg but is
    now separate of the call handling logic.

http://cgit.osmocom.org/osmo-sip-connector/commit/?id=b723cceee964ba57c23be30c7f736dece04b2ce3

commit b723cceee964ba57c23be30c7f736dece04b2ce3
Author: Holger Hans Peter Freyther <holger at moiji-mobile.com>
Date:   Sat Mar 26 19:51:33 2016 +0100

    call: Create a call with a SIP leg
    
    Clone the MNCC code and create a call with a SIP leg.

http://cgit.osmocom.org/osmo-sip-connector/commit/?id=586abf9f0fba385de1a8b22376a870a556909d97

commit 586abf9f0fba385de1a8b22376a870a556909d97
Author: Holger Hans Peter Freyther <holger at moiji-mobile.com>
Date:   Sat Mar 26 19:44:03 2016 +0100

    sip: Handle incoming message and screen the SDP file
    
    Check if the SDP file has any codec potentially supported by GSM.
    The topic of codec selection is a complicated one and we will not
    support it correctly in the beginning.

http://cgit.osmocom.org/osmo-sip-connector/commit/?id=6f039c158deb36a7995c828460bad49f0b9db642

commit 6f039c158deb36a7995c828460bad49f0b9db642
Author: Holger Hans Peter Freyther <holger at moiji-mobile.com>
Date:   Sat Mar 26 18:55:33 2016 +0100

    sip: Mark call as MT call in preparation for incoming calls

http://cgit.osmocom.org/osmo-sip-connector/commit/?id=b927a0bbdebab81c108059ad6d6a0a812fff2b75

commit b927a0bbdebab81c108059ad6d6a0a812fff2b75
Author: Holger Hans Peter Freyther <holger at moiji-mobile.com>
Date:   Sat Mar 26 18:52:22 2016 +0100

    call: It is call_create and not sip_create
    
    It is not creating anything related to sip at all. Remove the
    prefix in preparation to creating a call from a sip_call_leg.

-----------------------------------------------------------------------

Summary of changes:
 src/Makefile.am |   3 +-
 src/app.c       |   2 +-
 src/call.c      |  26 ++++++++-
 src/call.h      |  14 ++++-
 src/mncc.c      |   2 +-
 src/sdp.c       | 161 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 src/sdp.h       |  10 ++++
 src/sip.c       | 155 +++++++++++++++++++++++++++--------------------------
 8 files changed, 293 insertions(+), 80 deletions(-)
 create mode 100644 src/sdp.c
 create mode 100644 src/sdp.h


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