Merge branch 'master' of git+ssh://repos-sflphone-git@git.sflphone.org/var/repos/sflphone/git/sflphone

it. Some values were hard-coded for testing purposes. Also please be
careful about the local/published addresses and port that are used.

Preliminary tests with Freeswitch show that the SSL connection can be
established but if fails because of "invalid transport "SIP/2.0/tls".

gtk-ui. Values are not saved back yet.

Since STUN is not suitable for TCP transport, this information must be
provided manually by the user. Otherwise, user must configure
manual (static) traversal on her/his firewall.

Not integrated yet under the UI.

starting with that scheme.

commit adds the ability to create the right sip uri depending on the
transport selected.

This is a BACKUP kind of commit. The code is compilable, but yet UNTESTED !

was changed in order to be able to compile against pjsip with TLS enabled.

of SRTP_ENABLE.

to provide a default value in case a given key is not present in a map.

by providing default values when missing from request. This might have break
some features however: boolean values are represented as "0/1", transformed
into "true/false" strings, and converted back again ... We now simply use
"true/false" strings.

for SIP. The dialogue has no effect on the daemon at the moment.

prone to crashes.

when an error occur during zrtp negotiation.

an error occured with libzrtpcpp. Also, exceptions in AudioRtpFactory are
now derived from some more specialized types.

when using the assistant.

registered types of accounts. ZRTP key exchange number was not in
accordance with the index from the daemon.

compilation information for pjsip-1.0.3.

profile in sflphone-client-gnome.

feature.

call. Also, the pulseaudio layer is no more drained because of the time (useless)
it could take in some circumstances.