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codec_negotiation: Implement outgoing_call_offer_pref

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Based on this new endpoint setting, a joint list of preferred codecs
between those received from the Asterisk core (remote), and those
specified in the endpoint's "allow" parameter (local) is created and
is used to create the outgoing SDP offer.

* Add outgoing_call_offer_pref to pjsip_configuration (endpoint)

* Add "call_direction" to res_pjsip_session.

* Update pjsip_session_caps.c to make the functions more generic
  so they could be used for both incoming and outgoing.

* Update ast_sip_session_create_outgoing to create the
  pending_media_state->topology with the results of
  ast_sip_session_create_joint_call_stream().

* The endpoint "preferred_codec_only" option now automatically sets
  AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref.

* A helper function ast_stream_get_format_count() was added to
  streams to return the current count of formats.

ASTERISK-28777

Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437
This commit is contained in:
George Joseph
2020-03-13 13:40:46 -06:00
committed by Friendly Automation
parent 57a457c26c
commit 2ee455958e
13 changed files with 408 additions and 297 deletions
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93
res/res_pjsip.c
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@@ -923,25 +923,75 @@
</para></description>
</configOption>
<configOption name="preferred_codec_only" default="no">
<synopsis>Respond to a SIP invite with the single most preferred codec rather than advertising all joint codec capabilities. This limits the other side's codec choice to exactly what we prefer.</synopsis>
<synopsis>Respond to a SIP invite with the single most preferred codec (DEPRECATED)</synopsis>
<description><para>Respond to a SIP invite with the single most preferred codec
rather than advertising all joint codec capabilities. This limits the other side's codec
choice to exactly what we prefer.</para>
<warning><para>This option has been deprecated in favor of
<literal>incoming_call_offer_pref</literal>. Setting both options is unsupported.</para>
</warning>
</description>
<see-also>
<ref type="configOption">incoming_call_offer_pref</ref>
</see-also>
</configOption>
<configOption name="incoming_call_offer_pref" default="local">
<synopsis>After receiving an incoming offer create a list of preferred codecs between
those received in the SDP offer, and those specified in endpoint configuration.</synopsis>
<synopsis>Preferences for selecting codecs for an incoming call.</synopsis>
<description>
<note><para>This list will consist of only those codecs found in both.</para></note>
<para>Based on this setting, a joint list of preferred codecs between those
received in an incoming SDP offer (remote), and those specified in the
endpoint's "allow" parameter (local) es created and is passed to the Asterisk
core. </para>
<note><para>This list will consist of only those codecs found in both lists.</para></note>
<enumlist>
<enum name="local"><para>
Order by the endpoint configuration allow line (default)
Include all codecs in the local list that are also in the remote list
preserving the local order. (default).
</para></enum>
<enum name="local_single"><para>
Order by the endpoint configuration allow line, but the list will only contain the first, or 'top' item
<enum name="local_first"><para>
Include only the first codec in the local list that is also in the remote list.
</para></enum>
<enum name="remote"><para>
Order by what is received in the SDP offer
Include all codecs in the remote list that are also in the local list
preserving the remote order.
</para></enum>
<enum name="remote_single"><para>
Order by what is received in the SDP offer, but the list will only contain the first, or 'top' item
<enum name="remote_first"><para>
Include only the first codec in the remote list that is also in the local list.
</para></enum>
</enumlist>
</description>
</configOption>
<configOption name="outgoing_call_offer_pref" default="local">
<synopsis>Preferences for selecting codecs for an outgoing call.</synopsis>
<description>
<para>Based on this setting, a joint list of preferred codecs between
those received from the Asterisk core (remote), and those specified in
the endpoint's "allow" parameter (local) is created and is used to create
the outgoing SDP offer.</para>
<enumlist>
<enum name="local"><para>
Include all codecs in the local list that are also in the remote list
preserving the local order.
</para></enum>
<enum name="local_merge"><para>
Include all codecs in BOTH lists preserving the local order.
Remote codecs not in the local list will be placed at the end
of the joint list.
</para></enum>
<enum name="local_first"><para>
Include only the first codec in the local list.
</para></enum>
<enum name="remote"><para>
Include all codecs in the remote list that are also in the local list
preserving the remote order. (default)
</para></enum>
<enum name="remote_merge"><para>
Include all codecs in BOTH lists preserving the remote order.
Local codecs not in the remote list will be placed at the end
of the joint list.
</para></enum>
<enum name="remote_first"><para>
Include only the first codec in the remote list.
</para></enum>
</enumlist>
</description>
@@ -5044,6 +5094,29 @@ int ast_sip_str_to_dtmf(const char * dtmf_mode)
return result;
}
const char *ast_sip_call_codec_pref_to_str(struct ast_flags pref)
{
const char *value;
if (ast_sip_call_codec_pref_test(pref, LOCAL) && ast_sip_call_codec_pref_test(pref, INTERSECT) && ast_sip_call_codec_pref_test(pref, ALL)) {
value = "local";
} else if (ast_sip_call_codec_pref_test(pref, LOCAL) && ast_sip_call_codec_pref_test(pref, UNION) && ast_sip_call_codec_pref_test(pref, ALL)) {
value = "local_merge";
} else if (ast_sip_call_codec_pref_test(pref, LOCAL) && ast_sip_call_codec_pref_test(pref, INTERSECT) && ast_sip_call_codec_pref_test(pref, FIRST)) {
value = "local_first";
} else if (ast_sip_call_codec_pref_test(pref, REMOTE) && ast_sip_call_codec_pref_test(pref, INTERSECT) && ast_sip_call_codec_pref_test(pref, ALL)) {
value = "remote";
} else if (ast_sip_call_codec_pref_test(pref, REMOTE) && ast_sip_call_codec_pref_test(pref, UNION) && ast_sip_call_codec_pref_test(pref, ALL)) {
value = "remote_merge";
} else if (ast_sip_call_codec_pref_test(pref, REMOTE) && ast_sip_call_codec_pref_test(pref, UNION) && ast_sip_call_codec_pref_test(pref, FIRST)) {
value = "remote_first";
} else {
value = "unknown";
}
return value;
}
/*!
* \brief Set name and number information on an identity header.
*