[asterisk-dev] [asterisk-commits] file: trunk r427112 - in /trunk: channels/ channels/pjsip/ configs/samples/ c...
Matthew Jordan
mjordan at digium.com
Mon Nov 3 09:00:52 CST 2014
Should have caught this in the review, but do you mind updating the
CHANGES file for this as well? :-)
On Mon, Nov 3, 2014 at 8:45 AM, SVN commits to the Asterisk project
<asterisk-commits at lists.digium.com> wrote:
> Author: file
> Date: Mon Nov 3 08:45:01 2014
> New Revision: 427112
>
> URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=427112
> Log:
> chan_pjsip: Add support for passing hold and unhold requests through.
>
> This change adds an option, moh_passthrough, that when enabled will pass
> hold and unhold requests through using a SIP re-invite. When placing on
> hold a re-invite with sendonly will be sent and when taking off hold a
> re-invite with sendrecv will be sent. This allows remote servers to handle
> the musiconhold instead of the local Asterisk instance being responsible.
>
> Review: https://reviewboard.asterisk.org/r/4103/
>
> Added:
> trunk/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py (with props)
> Modified:
> trunk/channels/chan_pjsip.c
> trunk/channels/pjsip/dialplan_functions.c
> trunk/configs/samples/pjsip.conf.sample
> trunk/include/asterisk/res_pjsip.h
> trunk/include/asterisk/res_pjsip_session.h
> trunk/res/res_pjsip.c
> trunk/res/res_pjsip/pjsip_configuration.c
> trunk/res/res_pjsip_sdp_rtp.c
>
> Modified: trunk/channels/chan_pjsip.c
> URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_pjsip.c?view=diff&rev=427112&r1=427111&r2=427112
> ==============================================================================
> --- trunk/channels/chan_pjsip.c (original)
> +++ trunk/channels/chan_pjsip.c Mon Nov 3 08:45:01 2014
> @@ -1097,6 +1097,39 @@
> return 0;
> }
>
> +/*! \brief Callback which changes the value of locally held on the media stream */
> +static int local_hold_set_state(void *obj, void *arg, int flags)
> +{
> + struct ast_sip_session_media *session_media = obj;
> + unsigned int *held = arg;
> +
> + session_media->locally_held = *held;
> +
> + return 0;
> +}
> +
> +/*! \brief Update local hold state and send a re-INVITE with the new SDP */
> +static int remote_send_hold_refresh(struct ast_sip_session *session, unsigned int held)
> +{
> + ao2_callback(session->media, OBJ_NODATA, local_hold_set_state, &held);
> + ast_sip_session_refresh(session, NULL, NULL, NULL, AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1);
> + ao2_ref(session, -1);
> +
> + return 0;
> +}
> +
> +/*! \brief Update local hold state to be held */
> +static int remote_send_hold(void *data)
> +{
> + return remote_send_hold_refresh(data, 1);
> +}
> +
> +/*! \brief Update local hold state to be unheld */
> +static int remote_send_unhold(void *data)
> +{
> + return remote_send_hold_refresh(data, 0);
> +}
> +
> /*! \brief Function called by core to ask the channel to indicate some sort of condition */
> static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
> {
> @@ -1219,7 +1252,15 @@
> device_buf = alloca(device_buf_size);
> ast_channel_get_device_name(ast, device_buf, device_buf_size);
> ast_devstate_changed_literal(AST_DEVICE_ONHOLD, 1, device_buf);
> - ast_moh_start(ast, data, NULL);
> + if (!channel->session->endpoint->moh_passthrough) {
> + ast_moh_start(ast, data, NULL);
> + } else {
> + if (ast_sip_push_task(channel->session->serializer, remote_send_hold, ao2_bump(channel->session))) {
> + ast_log(LOG_WARNING, "Could not queue task to remotely put session '%s' on hold with endpoint '%s'\n",
> + ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint));
> + ao2_ref(channel->session, -1);
> + }
> + }
> break;
> case AST_CONTROL_UNHOLD:
> chan_pjsip_remove_hold(ast_channel_uniqueid(ast));
> @@ -1227,7 +1268,15 @@
> device_buf = alloca(device_buf_size);
> ast_channel_get_device_name(ast, device_buf, device_buf_size);
> ast_devstate_changed_literal(AST_DEVICE_UNKNOWN, 1, device_buf);
> - ast_moh_stop(ast);
> + if (!channel->session->endpoint->moh_passthrough) {
> + ast_moh_stop(ast);
> + } else {
> + if (ast_sip_push_task(channel->session->serializer, remote_send_unhold, ao2_bump(channel->session))) {
> + ast_log(LOG_WARNING, "Could not queue task to remotely take session '%s' off hold with endpoint '%s'\n",
> + ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint));
> + ao2_ref(channel->session, -1);
> + }
> + }
> break;
> case AST_CONTROL_SRCUPDATE:
> break;
>
> Modified: trunk/channels/pjsip/dialplan_functions.c
> URL: http://svnview.digium.com/svn/asterisk/trunk/channels/pjsip/dialplan_functions.c?view=diff&rev=427112&r1=427111&r2=427112
> ==============================================================================
> --- trunk/channels/pjsip/dialplan_functions.c (original)
> +++ trunk/channels/pjsip/dialplan_functions.c Mon Nov 3 08:45:01 2014
> @@ -434,7 +434,7 @@
> } else if (!strcmp(type, "secure")) {
> snprintf(buf, buflen, "%d", media->srtp ? 1 : 0);
> } else if (!strcmp(type, "hold")) {
> - snprintf(buf, buflen, "%d", media->held ? 1 : 0);
> + snprintf(buf, buflen, "%d", media->remotely_held ? 1 : 0);
> } else {
> ast_log(AST_LOG_WARNING, "Unknown type field '%s' specified for 'rtp' information\n", type);
> return -1;
>
> Modified: trunk/configs/samples/pjsip.conf.sample
> URL: http://svnview.digium.com/svn/asterisk/trunk/configs/samples/pjsip.conf.sample?view=diff&rev=427112&r1=427111&r2=427112
> ==============================================================================
> --- trunk/configs/samples/pjsip.conf.sample (original)
> +++ trunk/configs/samples/pjsip.conf.sample Mon Nov 3 08:45:01 2014
> @@ -616,6 +616,8 @@
> ; (default: "user")
> ;mailboxes= ; Mailbox es to be associated with (default: "")
> ;moh_suggest=default ; Default Music On Hold class (default: "default")
> +;moh_passthrough=yes ; Pass Music On Hold through using SIP re-invites with sendonly
> + ; when placing on hold and sendrecv when taking off hold
> ;outbound_auth= ; Authentication object used for outbound requests (default:
> ; "")
> ;outbound_proxy= ; Proxy through which to send requests a full SIP URI
>
> Added: trunk/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py
> URL: http://svnview.digium.com/svn/asterisk/trunk/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py?view=auto&rev=427112
> ==============================================================================
> --- trunk/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py (added)
> +++ trunk/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py Mon Nov 3 08:45:01 2014
> @@ -1,0 +1,30 @@
> +"""Add moh_passthrough option to pjsip
> +
> +Revision ID: 339e1dfa644d
> +Revises: 1443687dda65
> +Create Date: 2014-10-21 14:55:34.197448
> +
> +"""
> +
> +# revision identifiers, used by Alembic.
> +revision = '339e1dfa644d'
> +down_revision = '1443687dda65'
> +
> +from alembic import op
> +import sqlalchemy as sa
> +from sqlalchemy.dialects.postgresql import ENUM
> +
> +YESNO_NAME = 'yesno_values'
> +YESNO_VALUES = ['yes', 'no']
> +
> +def upgrade():
> + ############################# Enums ##############################
> +
> + # yesno_values have already been created, so use postgres enum object
> + # type to get around "already created" issue - works okay with mysql
> + yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False)
> +
> + op.add_column('ps_endpoints', sa.Column('moh_passthrough', yesno_values))
> +
> +def downgrade():
> + op.drop_column('ps_endpoints', 'moh_passthrough')
>
> Propchange: trunk/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py
> ------------------------------------------------------------------------------
> svn:eol-style = native
>
> Propchange: trunk/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py
> ------------------------------------------------------------------------------
> svn:keywords = Author Date Id Revision
>
> Propchange: trunk/contrib/ast-db-manage/config/versions/339e1dfa644d_add_moh_passthrough_option_to_pjsip.py
> ------------------------------------------------------------------------------
> svn:mime-type = text/plain
>
> Modified: trunk/include/asterisk/res_pjsip.h
> URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/res_pjsip.h?view=diff&rev=427112&r1=427111&r2=427112
> ==============================================================================
> --- trunk/include/asterisk/res_pjsip.h (original)
> +++ trunk/include/asterisk/res_pjsip.h Mon Nov 3 08:45:01 2014
> @@ -609,6 +609,8 @@
> struct ast_variable *channel_vars;
> /*! Whether to place a 'user=phone' parameter into the request URI if user is a number */
> unsigned int usereqphone;
> + /*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */
> + unsigned int moh_passthrough;
> };
>
> /*!
>
> Modified: trunk/include/asterisk/res_pjsip_session.h
> URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/res_pjsip_session.h?view=diff&rev=427112&r1=427111&r2=427112
> ==============================================================================
> --- trunk/include/asterisk/res_pjsip_session.h (original)
> +++ trunk/include/asterisk/res_pjsip_session.h Mon Nov 3 08:45:01 2014
> @@ -75,8 +75,10 @@
> struct ast_sdp_srtp *srtp;
> /*! \brief The media transport in use for this stream */
> pj_str_t transport;
> - /*! \brief Stream is on hold */
> - unsigned int held:1;
> + /*! \brief Stream is on hold by remote side */
> + unsigned int remotely_held:1;
> + /*! \brief Stream is on hold by local side */
> + unsigned int locally_held:1;
> /*! \brief Stream type this session media handles */
> char stream_type[1];
> };
>
> Modified: trunk/res/res_pjsip.c
> URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip.c?view=diff&rev=427112&r1=427111&r2=427112
> ==============================================================================
> --- trunk/res/res_pjsip.c (original)
> +++ trunk/res/res_pjsip.c Mon Nov 3 08:45:01 2014
> @@ -576,6 +576,9 @@
> <configOption name="user_eq_phone" default="no">
> <synopsis>Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number</synopsis>
> </configOption>
> + <configOption name="moh_passthrough" default="no">
> + <synopsis>Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side</synopsis>
> + </configOption>
> <configOption name="sdp_owner" default="-">
> <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis>
> </configOption>
> @@ -1559,6 +1562,9 @@
> </parameter>
> <parameter name="UserEqPhone">
> <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='user_eq_phone']/synopsis/node())"/></para>
> + </parameter>
> + <parameter name="MohPassthrough">
> + <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='moh_passthrough']/synopsis/node())"/></para>
> </parameter>
> <parameter name="SdpOwner">
> <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_owner']/synopsis/node())"/></para>
>
> Modified: trunk/res/res_pjsip/pjsip_configuration.c
> URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip/pjsip_configuration.c?view=diff&rev=427112&r1=427111&r2=427112
> ==============================================================================
> --- trunk/res/res_pjsip/pjsip_configuration.c (original)
> +++ trunk/res/res_pjsip/pjsip_configuration.c Mon Nov 3 08:45:01 2014
> @@ -1733,6 +1733,7 @@
> ast_sorcery_object_field_register(sip_sorcery, "endpoint", "record_off_feature", "automixmon", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, info.recording.offfeature));
> ast_sorcery_object_field_register(sip_sorcery, "endpoint", "allow_transfer", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, allowtransfer));
> ast_sorcery_object_field_register(sip_sorcery, "endpoint", "user_eq_phone", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, usereqphone));
> + ast_sorcery_object_field_register(sip_sorcery, "endpoint", "moh_passthrough", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, moh_passthrough));
> ast_sorcery_object_field_register(sip_sorcery, "endpoint", "sdp_owner", "-", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, media.sdpowner));
> ast_sorcery_object_field_register(sip_sorcery, "endpoint", "sdp_session", "Asterisk", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, media.sdpsession));
> ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "tos_audio", "0", tos_handler, tos_audio_to_str, NULL, 0, 0);
>
> Modified: trunk/res/res_pjsip_sdp_rtp.c
> URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_pjsip_sdp_rtp.c?view=diff&rev=427112&r1=427111&r2=427112
> ==============================================================================
> --- trunk/res/res_pjsip_sdp_rtp.c (original)
> +++ trunk/res/res_pjsip_sdp_rtp.c Mon Nov 3 08:45:01 2014
> @@ -887,6 +887,7 @@
> static const pj_str_t STR_IP4 = { "IP4", 3};
> static const pj_str_t STR_IP6 = { "IP6", 3};
> static const pj_str_t STR_SENDRECV = { "sendrecv", 8 };
> + static const pj_str_t STR_SENDONLY = { "sendonly", 8 };
> pjmedia_sdp_media *media;
> char hostip[PJ_INET6_ADDRSTRLEN+2];
> struct ast_sockaddr addr;
> @@ -1046,7 +1047,7 @@
>
> /* Add the sendrecv attribute - we purposely don't keep track because pjmedia-sdp will automatically change our offer for us */
> attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr);
> - attr->name = STR_SENDRECV;
> + attr->name = !session_media->locally_held ? STR_SENDRECV : STR_SENDONLY;
> media->attr[media->attr_count++] = attr;
>
> /* Add the media stream to the SDP */
> @@ -1122,18 +1123,18 @@
> if (ast_sockaddr_isnull(addrs) ||
> ast_sockaddr_is_any(addrs) ||
> pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL)) {
> - if (!session_media->held) {
> + if (!session_media->remotely_held) {
> /* The remote side has put us on hold */
> ast_queue_hold(session->channel, session->endpoint->mohsuggest);
> ast_rtp_instance_stop(session_media->rtp);
> ast_queue_frame(session->channel, &ast_null_frame);
> - session_media->held = 1;
> - }
> - } else if (session_media->held) {
> + session_media->remotely_held = 1;
> + }
> + } else if (session_media->remotely_held) {
> /* The remote side has taken us off hold */
> ast_queue_unhold(session->channel);
> ast_queue_frame(session->channel, &ast_null_frame);
> - session_media->held = 0;
> + session_media->remotely_held = 0;
> }
>
> return 1;
>
>
> --
> _____________________________________________________________________
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--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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