Bug#708122: ITP: pjsip -- VoIP library implementing SIP and related protocols

[Tzafrir Cohen]

Package: wnpp
Severity: wishlist
Owner: Tzafrir Cohen <tzafrir at debian.org>

* Package name    : pjsip
  Version         : 2.1.0.0.ast20130312
  Upstream Author : http://www.teluu.com/
* URL             : http://pjsip.org/
* License         : GPL2+ (with OpenSSL exceptions)
  Programming Lang: C
  Description     : VoIP library implementing SIP and related protocols

 multimedia communication library implementing standard based protocols
 such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling
 protocol (SIP) with multimedia framework and NAT traversal functionality
 into high level multimedia communication API that is portable and
 suitable for almost any type of systems ranging from desktops, embedded
 systems, to mobile handsets.

PJSIP (a.k.a PJProject. pjsip may refer to only one of its components,
implementing the SIP protocol) is normally built as a static library and
embedded in projects. It is currently embedded in sflphone (any other
package?).

Asterisk 11 uses an embedded copy of pjsip for some extra features in
the SIP channel driver. Asterisk >= 12 (current trunk) will use it as
the base for its SIP channel driver (and possibly more things). In order
to avoid an embedded copy of the library, the developers of Asterisk
maintain their own copy of pjsip which is dynamically-linked[1]. This
packaging is based on that copy. According to them this patch set should
be merged back to the pjsip maintainers by October (when Asterisk 12 is
released).

Initial packaging could be found at [2] (named "pjproject"). Packages
built from it: at [3]. Specifically, I'm not exactly sure if we need
more than the single -dev package / component. See [4].

[1] https://github.com/asterisk/pjproject
[2] http://git.tzafrir.org.il/?p=pjproject.git;a=summary
[3] http://tzafrir.org.il/~tzafrir/ast_11/
[4] http://lists.digium.com/pipermail/asterisk-dev/2013-April/059577.html


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