This patch addresses the following problems: * ari/resource_channels: In ARI, we currently create a format capability structure of SLIN and apply it to the new channel being created. This was originally done when the PBX core was used to create the channel, as there was a condition where a newly created channel could be created without any formats. Unfortunately, now that the Dial API is being used, this has two drawbacks: (a) SLIN, while it will ensure audio will flows, can cause a lot of needless transcodings to occur, particularly when a Local channel is created to the dialplan. When no format capabilities are available, the Dial API handles this better by handing all audio formats to the requsted channels. As such, we defer to that API to provide the format capabilities. (b) If a channel (requester) is causing this channel to be created, we currently don't use its format capabilities as we are passing in our own. However, the Dial API will use the requester channel's formats if none are passed into it, and the requester channel exists and has format capabilities. This is the "best" scenario, as it is the most likely to create a media path that minimizes transcoding. Fixing this simply entails removing the providing of the format capabilities structure to the Dial API. * chan_pjsip: Rather than blindly picking the first format in the format capability structure - which actually *can* be a video or text format - we select an audio format, and only pick the first format if that fails. That minimizes the weird scenario where we attempt to transcode between video/audio. * res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure. Since ast_request already limits us down to one format capability once the format capabilities are passed along, there's no reason to squelch it here. * channel: Fixed a comment. The reason we have to minimize our requested format capabilities down to a single format is due to Asterisk's inability to convey the format to be used back "up" a channel chain. Consider the following: PJSIP/A => L;1 <=> L;2 => PJSIP/B g,u,a g,u,a g,u,a u That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local channel has inherited those format capabilities down the line; PJSIP/B supports only ulaw. According to these format capabilities, ulaw is acceptable and should be selected across all the channels, and no transcoding should occur. However, there is no way to convey this: when L;2 and PJSIP/B are put into a bridge, we will select ulaw, but that is not conveyed to PJSIP/A and L;1. Thus, we end up with: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B g g X u u Which causes g722 to be written to PJSIP/B. Even if we can convey the 'ulaw' choice back up the chain (which through some severe hacking in Local channels was accomplished), such that the chain looks like: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B u u u u We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back with only 'ulaw'. This results in all the channel structures being set up correctly, but PJSIP/A *still* sending g722 and causing the chain to fall apart. There's a lot of difficulty just in setting this up, as there are numerous race conditions in the act of bridging, and no clean mechanism to pass the selected format backwards down an established channel chain. As such, the best that can be done at this point in time is clarifying the comment. Review: https://reviewboard.asterisk.org/r/4434/ ASTERISK-24812 #close Reported by: Matt Jordan ........ Merged revisions 432195 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1345 lines
44 KiB
C
1345 lines
44 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2013, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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* Kevin Harwell <kharwell@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \author Joshua Colp <jcolp@digium.com>
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*
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* \brief SIP SDP media stream handling
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*/
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/*** MODULEINFO
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<depend>pjproject</depend>
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<depend>res_pjsip</depend>
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<depend>res_pjsip_session</depend>
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include <pjsip.h>
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#include <pjsip_ua.h>
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#include <pjmedia.h>
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#include <pjlib.h>
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/module.h"
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#include "asterisk/format.h"
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#include "asterisk/format_cap.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/netsock2.h"
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#include "asterisk/channel.h"
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#include "asterisk/causes.h"
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#include "asterisk/sched.h"
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#include "asterisk/acl.h"
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#include "asterisk/sdp_srtp.h"
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#include "asterisk/res_pjsip.h"
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#include "asterisk/res_pjsip_session.h"
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/*! \brief Scheduler for RTCP purposes */
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static struct ast_sched_context *sched;
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/*! \brief Address for IPv4 RTP */
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static struct ast_sockaddr address_ipv4;
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/*! \brief Address for IPv6 RTP */
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static struct ast_sockaddr address_ipv6;
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static const char STR_AUDIO[] = "audio";
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static const int FD_AUDIO = 0;
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static const char STR_VIDEO[] = "video";
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static const int FD_VIDEO = 2;
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/*! \brief Retrieves an ast_format_type based on the given stream_type */
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static enum ast_media_type stream_to_media_type(const char *stream_type)
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{
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if (!strcasecmp(stream_type, STR_AUDIO)) {
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return AST_MEDIA_TYPE_AUDIO;
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} else if (!strcasecmp(stream_type, STR_VIDEO)) {
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return AST_MEDIA_TYPE_VIDEO;
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}
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return 0;
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}
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/*! \brief Get the starting descriptor for a media type */
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static int media_type_to_fdno(enum ast_media_type media_type)
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{
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switch (media_type) {
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case AST_MEDIA_TYPE_AUDIO: return FD_AUDIO;
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case AST_MEDIA_TYPE_VIDEO: return FD_VIDEO;
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case AST_MEDIA_TYPE_TEXT:
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case AST_MEDIA_TYPE_UNKNOWN:
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case AST_MEDIA_TYPE_IMAGE: break;
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}
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return -1;
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}
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/*! \brief Remove all other cap types but the one given */
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static void format_cap_only_type(struct ast_format_cap *caps, enum ast_media_type media_type)
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{
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int i = 0;
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while (i <= AST_MEDIA_TYPE_TEXT) {
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if (i != media_type && i != AST_MEDIA_TYPE_UNKNOWN) {
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ast_format_cap_remove_by_type(caps, i);
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}
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i += 1;
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}
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}
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/*! \brief Internal function which creates an RTP instance */
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static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media, unsigned int ipv6)
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{
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struct ast_rtp_engine_ice *ice;
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if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->media.rtp.engine, sched, ipv6 ? &address_ipv6 : &address_ipv4, NULL))) {
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ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n", session->endpoint->media.rtp.engine);
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return -1;
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}
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ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RTCP, 1);
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ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_NAT, session->endpoint->media.rtp.symmetric);
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if (!session->endpoint->media.rtp.ice_support && (ice = ast_rtp_instance_get_ice(session_media->rtp))) {
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ice->stop(session_media->rtp);
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}
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if (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) {
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ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
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ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 1);
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} else if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
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ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
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}
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if (!strcmp(session_media->stream_type, STR_AUDIO) &&
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(session->endpoint->media.tos_audio || session->endpoint->media.cos_video)) {
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ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio,
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session->endpoint->media.cos_audio, "SIP RTP Audio");
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} else if (!strcmp(session_media->stream_type, STR_VIDEO) &&
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(session->endpoint->media.tos_video || session->endpoint->media.cos_video)) {
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ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video,
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session->endpoint->media.cos_video, "SIP RTP Video");
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}
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return 0;
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}
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static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp_media *stream, struct ast_rtp_codecs *codecs)
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{
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pjmedia_sdp_attr *attr;
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pjmedia_sdp_rtpmap *rtpmap;
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pjmedia_sdp_fmtp fmtp;
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struct ast_format *format;
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int i, num = 0;
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char name[256];
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char media[20];
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char fmt_param[256];
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ast_rtp_codecs_payloads_initialize(codecs);
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/* Iterate through provided formats */
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for (i = 0; i < stream->desc.fmt_count; ++i) {
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/* The payload is kept as a string for things like t38 but for video it is always numerical */
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ast_rtp_codecs_payloads_set_m_type(codecs, NULL, pj_strtoul(&stream->desc.fmt[i]));
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/* Look for the optional rtpmap attribute */
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if (!(attr = pjmedia_sdp_media_find_attr2(stream, "rtpmap", &stream->desc.fmt[i]))) {
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continue;
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}
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/* Interpret the attribute as an rtpmap */
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if ((pjmedia_sdp_attr_to_rtpmap(session->inv_session->pool_prov, attr, &rtpmap)) != PJ_SUCCESS) {
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continue;
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}
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ast_copy_pj_str(name, &rtpmap->enc_name, sizeof(name));
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ast_copy_pj_str(media, (pj_str_t*)&stream->desc.media, sizeof(media));
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ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL, pj_strtoul(&stream->desc.fmt[i]),
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media, name, 0, rtpmap->clock_rate);
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/* Look for an optional associated fmtp attribute */
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if (!(attr = pjmedia_sdp_media_find_attr2(stream, "fmtp", &rtpmap->pt))) {
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continue;
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}
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if ((pjmedia_sdp_attr_get_fmtp(attr, &fmtp)) == PJ_SUCCESS) {
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sscanf(pj_strbuf(&fmtp.fmt), "%d", &num);
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if ((format = ast_rtp_codecs_get_payload_format(codecs, num))) {
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struct ast_format *format_parsed;
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ast_copy_pj_str(fmt_param, &fmtp.fmt_param, sizeof(fmt_param));
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format_parsed = ast_format_parse_sdp_fmtp(format, fmt_param);
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if (format_parsed) {
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ast_rtp_codecs_payload_replace_format(codecs, num, format_parsed);
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ao2_ref(format_parsed, -1);
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}
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ao2_ref(format, -1);
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}
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}
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}
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/* Get the packetization, if it exists */
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if ((attr = pjmedia_sdp_media_find_attr2(stream, "ptime", NULL))) {
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unsigned long framing = pj_strtoul(pj_strltrim(&attr->value));
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if (framing && session->endpoint->media.rtp.use_ptime) {
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ast_rtp_codecs_set_framing(codecs, framing);
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}
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}
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}
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static int set_caps(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
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const struct pjmedia_sdp_media *stream)
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{
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RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
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RAII_VAR(struct ast_format_cap *, peer, NULL, ao2_cleanup);
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RAII_VAR(struct ast_format_cap *, joint, NULL, ao2_cleanup);
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enum ast_media_type media_type = stream_to_media_type(session_media->stream_type);
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struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;
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int fmts = 0;
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int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
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ast_format_cap_count(session->direct_media_cap);
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if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
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!(peer = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
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!(joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
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ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n", session_media->stream_type);
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return -1;
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}
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/* get the endpoint capabilities */
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if (direct_media_enabled) {
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ast_format_cap_get_compatible(session->endpoint->media.codecs, session->direct_media_cap, caps);
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format_cap_only_type(caps, media_type);
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} else {
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ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, media_type);
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}
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/* get the capabilities on the peer */
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get_codecs(session, stream, &codecs);
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ast_rtp_codecs_payload_formats(&codecs, peer, &fmts);
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/* get the joint capabilities between peer and endpoint */
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ast_format_cap_get_compatible(caps, peer, joint);
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if (!ast_format_cap_count(joint)) {
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struct ast_str *usbuf = ast_str_alloca(256);
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struct ast_str *thembuf = ast_str_alloca(256);
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ast_rtp_codecs_payloads_destroy(&codecs);
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ast_log(LOG_NOTICE, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n",
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session_media->stream_type,
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ast_format_cap_get_names(caps, &usbuf),
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ast_format_cap_get_names(peer, &thembuf));
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return -1;
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}
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ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
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session_media->rtp);
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ast_format_cap_append_from_cap(session->req_caps, joint, AST_MEDIA_TYPE_UNKNOWN);
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if (session->channel) {
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ast_channel_lock(session->channel);
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/*
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* Apply the new formats to the channel, potentially changing
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* raw read/write formats and translation path while doing so.
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*/
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ast_channel_nativeformats_set(session->channel, joint);
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ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
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ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
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ast_channel_unlock(session->channel);
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}
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ast_rtp_codecs_payloads_destroy(&codecs);
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return 0;
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}
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static pjmedia_sdp_attr* generate_rtpmap_attr(pjmedia_sdp_media *media, pj_pool_t *pool, int rtp_code,
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int asterisk_format, struct ast_format *format, int code)
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{
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pjmedia_sdp_rtpmap rtpmap;
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pjmedia_sdp_attr *attr = NULL;
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char tmp[64];
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snprintf(tmp, sizeof(tmp), "%d", rtp_code);
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pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], tmp);
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rtpmap.pt = media->desc.fmt[media->desc.fmt_count - 1];
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rtpmap.clock_rate = ast_rtp_lookup_sample_rate2(asterisk_format, format, code);
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pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, 0));
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rtpmap.param.slen = 0;
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rtpmap.param.ptr = NULL;
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pjmedia_sdp_rtpmap_to_attr(pool, &rtpmap, &attr);
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return attr;
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}
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static pjmedia_sdp_attr* generate_fmtp_attr(pj_pool_t *pool, struct ast_format *format, int rtp_code)
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{
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struct ast_str *fmtp0 = ast_str_alloca(256);
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pj_str_t fmtp1;
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pjmedia_sdp_attr *attr = NULL;
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char *tmp;
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ast_format_generate_sdp_fmtp(format, rtp_code, &fmtp0);
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if (ast_str_strlen(fmtp0)) {
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tmp = ast_str_buffer(fmtp0) + ast_str_strlen(fmtp0) - 1;
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/* remove any carriage return line feeds */
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while (*tmp == '\r' || *tmp == '\n') --tmp;
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*++tmp = '\0';
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/* ast...generate gives us everything, just need value */
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tmp = strchr(ast_str_buffer(fmtp0), ':');
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if (tmp && tmp + 1) {
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fmtp1 = pj_str(tmp + 1);
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} else {
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fmtp1 = pj_str(ast_str_buffer(fmtp0));
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}
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attr = pjmedia_sdp_attr_create(pool, "fmtp", &fmtp1);
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}
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return attr;
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}
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/*! \brief Function which adds ICE attributes to a media stream */
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static void add_ice_to_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media, pj_pool_t *pool, pjmedia_sdp_media *media)
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{
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struct ast_rtp_engine_ice *ice;
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struct ao2_container *candidates;
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const char *username, *password;
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pj_str_t stmp;
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pjmedia_sdp_attr *attr;
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struct ao2_iterator it_candidates;
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struct ast_rtp_engine_ice_candidate *candidate;
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if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp)) ||
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!(candidates = ice->get_local_candidates(session_media->rtp))) {
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return;
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}
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if ((username = ice->get_ufrag(session_media->rtp))) {
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attr = pjmedia_sdp_attr_create(pool, "ice-ufrag", pj_cstr(&stmp, username));
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media->attr[media->attr_count++] = attr;
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}
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if ((password = ice->get_password(session_media->rtp))) {
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attr = pjmedia_sdp_attr_create(pool, "ice-pwd", pj_cstr(&stmp, password));
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media->attr[media->attr_count++] = attr;
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}
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it_candidates = ao2_iterator_init(candidates, 0);
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for (; (candidate = ao2_iterator_next(&it_candidates)); ao2_ref(candidate, -1)) {
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struct ast_str *attr_candidate = ast_str_create(128);
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ast_str_set(&attr_candidate, -1, "%s %u %s %d %s ", candidate->foundation, candidate->id, candidate->transport,
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candidate->priority, ast_sockaddr_stringify_addr_remote(&candidate->address));
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ast_str_append(&attr_candidate, -1, "%s typ ", ast_sockaddr_stringify_port(&candidate->address));
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switch (candidate->type) {
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case AST_RTP_ICE_CANDIDATE_TYPE_HOST:
|
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ast_str_append(&attr_candidate, -1, "host");
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break;
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case AST_RTP_ICE_CANDIDATE_TYPE_SRFLX:
|
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ast_str_append(&attr_candidate, -1, "srflx");
|
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break;
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case AST_RTP_ICE_CANDIDATE_TYPE_RELAYED:
|
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ast_str_append(&attr_candidate, -1, "relay");
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break;
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}
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|
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if (!ast_sockaddr_isnull(&candidate->relay_address)) {
|
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ast_str_append(&attr_candidate, -1, " raddr %s rport", ast_sockaddr_stringify_addr_remote(&candidate->relay_address));
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ast_str_append(&attr_candidate, -1, " %s", ast_sockaddr_stringify_port(&candidate->relay_address));
|
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}
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|
|
attr = pjmedia_sdp_attr_create(pool, "candidate", pj_cstr(&stmp, ast_str_buffer(attr_candidate)));
|
|
media->attr[media->attr_count++] = attr;
|
|
|
|
ast_free(attr_candidate);
|
|
}
|
|
|
|
ao2_iterator_destroy(&it_candidates);
|
|
ao2_ref(candidates, -1);
|
|
}
|
|
|
|
/*! \brief Function which processes ICE attributes in an audio stream */
|
|
static void process_ice_attributes(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
|
|
const struct pjmedia_sdp_session *remote, const struct pjmedia_sdp_media *remote_stream)
|
|
{
|
|
struct ast_rtp_engine_ice *ice;
|
|
const pjmedia_sdp_attr *attr;
|
|
char attr_value[256];
|
|
unsigned int attr_i;
|
|
|
|
/* If ICE support is not enabled or available exit early */
|
|
if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) {
|
|
return;
|
|
}
|
|
|
|
attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-ufrag", NULL);
|
|
if (!attr) {
|
|
attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-ufrag", NULL);
|
|
}
|
|
if (attr) {
|
|
ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
|
|
ice->set_authentication(session_media->rtp, attr_value, NULL);
|
|
} else {
|
|
return;
|
|
}
|
|
|
|
attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-pwd", NULL);
|
|
if (!attr) {
|
|
attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-pwd", NULL);
|
|
}
|
|
if (attr) {
|
|
ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
|
|
ice->set_authentication(session_media->rtp, NULL, attr_value);
|
|
} else {
|
|
return;
|
|
}
|
|
|
|
if (pjmedia_sdp_media_find_attr2(remote_stream, "ice-lite", NULL)) {
|
|
ice->ice_lite(session_media->rtp);
|
|
}
|
|
|
|
/* Find all of the candidates */
|
|
for (attr_i = 0; attr_i < remote_stream->attr_count; ++attr_i) {
|
|
char foundation[32], transport[32], address[PJ_INET6_ADDRSTRLEN + 1], cand_type[6], relay_address[PJ_INET6_ADDRSTRLEN + 1] = "";
|
|
unsigned int port, relay_port = 0;
|
|
struct ast_rtp_engine_ice_candidate candidate = { 0, };
|
|
|
|
attr = remote_stream->attr[attr_i];
|
|
|
|
/* If this is not a candidate line skip it */
|
|
if (pj_strcmp2(&attr->name, "candidate")) {
|
|
continue;
|
|
}
|
|
|
|
ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
|
|
|
|
if (sscanf(attr_value, "%31s %30u %31s %30u %46s %30u typ %5s %*s %23s %*s %30u", foundation, &candidate.id, transport,
|
|
(unsigned *)&candidate.priority, address, &port, cand_type, relay_address, &relay_port) < 7) {
|
|
/* Candidate did not parse properly */
|
|
continue;
|
|
}
|
|
|
|
candidate.foundation = foundation;
|
|
candidate.transport = transport;
|
|
|
|
ast_sockaddr_parse(&candidate.address, address, PARSE_PORT_FORBID);
|
|
ast_sockaddr_set_port(&candidate.address, port);
|
|
|
|
if (!strcasecmp(cand_type, "host")) {
|
|
candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_HOST;
|
|
} else if (!strcasecmp(cand_type, "srflx")) {
|
|
candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_SRFLX;
|
|
} else if (!strcasecmp(cand_type, "relay")) {
|
|
candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_RELAYED;
|
|
} else {
|
|
continue;
|
|
}
|
|
|
|
if (!ast_strlen_zero(relay_address)) {
|
|
ast_sockaddr_parse(&candidate.relay_address, relay_address, PARSE_PORT_FORBID);
|
|
}
|
|
|
|
if (relay_port) {
|
|
ast_sockaddr_set_port(&candidate.relay_address, relay_port);
|
|
}
|
|
|
|
ice->add_remote_candidate(session_media->rtp, &candidate);
|
|
}
|
|
|
|
ice->set_role(session_media->rtp, pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_TRUE ?
|
|
AST_RTP_ICE_ROLE_CONTROLLING : AST_RTP_ICE_ROLE_CONTROLLED);
|
|
ice->start(session_media->rtp);
|
|
}
|
|
|
|
/*! \brief figure out if media stream has crypto lines for sdes */
|
|
static int media_stream_has_crypto(const struct pjmedia_sdp_media *stream)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < stream->attr_count; i++) {
|
|
pjmedia_sdp_attr *attr;
|
|
|
|
/* check the stream for the required crypto attribute */
|
|
attr = stream->attr[i];
|
|
if (pj_strcmp2(&attr->name, "crypto")) {
|
|
continue;
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief figure out media transport encryption type from the media transport string */
|
|
static enum ast_sip_session_media_encryption get_media_encryption_type(pj_str_t transport,
|
|
const struct pjmedia_sdp_media *stream, unsigned int *optimistic)
|
|
{
|
|
RAII_VAR(char *, transport_str, ast_strndup(transport.ptr, transport.slen), ast_free);
|
|
|
|
*optimistic = 0;
|
|
|
|
if (strstr(transport_str, "UDP/TLS")) {
|
|
return AST_SIP_MEDIA_ENCRYPT_DTLS;
|
|
} else if (strstr(transport_str, "SAVP")) {
|
|
return AST_SIP_MEDIA_ENCRYPT_SDES;
|
|
} else if (media_stream_has_crypto(stream)) {
|
|
*optimistic = 1;
|
|
return AST_SIP_MEDIA_ENCRYPT_SDES;
|
|
} else {
|
|
return AST_SIP_MEDIA_ENCRYPT_NONE;
|
|
}
|
|
}
|
|
|
|
/*!
|
|
* \brief Checks whether the encryption offered in SDP is compatible with the endpoint's configuration
|
|
* \internal
|
|
*
|
|
* \param endpoint_encryption Media encryption configured for the endpoint
|
|
* \param stream pjmedia_sdp_media stream description
|
|
*
|
|
* \retval AST_SIP_MEDIA_TRANSPORT_INVALID on encryption mismatch
|
|
* \retval The encryption requested in the SDP
|
|
*/
|
|
static enum ast_sip_session_media_encryption check_endpoint_media_transport(
|
|
struct ast_sip_endpoint *endpoint,
|
|
const struct pjmedia_sdp_media *stream)
|
|
{
|
|
enum ast_sip_session_media_encryption incoming_encryption;
|
|
char transport_end = stream->desc.transport.ptr[stream->desc.transport.slen - 1];
|
|
unsigned int optimistic;
|
|
|
|
if ((transport_end == 'F' && !endpoint->media.rtp.use_avpf)
|
|
|| (transport_end != 'F' && endpoint->media.rtp.use_avpf)) {
|
|
return AST_SIP_MEDIA_TRANSPORT_INVALID;
|
|
}
|
|
|
|
incoming_encryption = get_media_encryption_type(stream->desc.transport, stream, &optimistic);
|
|
|
|
if (incoming_encryption == endpoint->media.rtp.encryption) {
|
|
return incoming_encryption;
|
|
}
|
|
|
|
if (endpoint->media.rtp.force_avp ||
|
|
endpoint->media.rtp.encryption_optimistic) {
|
|
return incoming_encryption;
|
|
}
|
|
|
|
/* If an optimistic offer has been made but encryption is not enabled consider it as having
|
|
* no offer of crypto at all instead of invalid so the session proceeds.
|
|
*/
|
|
if (optimistic) {
|
|
return AST_SIP_MEDIA_ENCRYPT_NONE;
|
|
}
|
|
|
|
return AST_SIP_MEDIA_TRANSPORT_INVALID;
|
|
}
|
|
|
|
static int setup_srtp(struct ast_sip_session_media *session_media)
|
|
{
|
|
if (!session_media->srtp) {
|
|
session_media->srtp = ast_sdp_srtp_alloc();
|
|
if (!session_media->srtp) {
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
if (!session_media->srtp->crypto) {
|
|
session_media->srtp->crypto = ast_sdp_crypto_alloc();
|
|
if (!session_media->srtp->crypto) {
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int setup_dtls_srtp(struct ast_sip_session *session,
|
|
struct ast_sip_session_media *session_media)
|
|
{
|
|
struct ast_rtp_engine_dtls *dtls;
|
|
|
|
if (!session->endpoint->media.rtp.dtls_cfg.enabled || !session_media->rtp) {
|
|
return -1;
|
|
}
|
|
|
|
dtls = ast_rtp_instance_get_dtls(session_media->rtp);
|
|
if (!dtls) {
|
|
return -1;
|
|
}
|
|
|
|
session->endpoint->media.rtp.dtls_cfg.suite = ((session->endpoint->media.rtp.srtp_tag_32) ? AST_AES_CM_128_HMAC_SHA1_32 : AST_AES_CM_128_HMAC_SHA1_80);
|
|
if (dtls->set_configuration(session_media->rtp, &session->endpoint->media.rtp.dtls_cfg)) {
|
|
ast_log(LOG_ERROR, "Attempted to set an invalid DTLS-SRTP configuration on RTP instance '%p'\n",
|
|
session_media->rtp);
|
|
return -1;
|
|
}
|
|
|
|
if (setup_srtp(session_media)) {
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static void apply_dtls_attrib(struct ast_sip_session_media *session_media,
|
|
pjmedia_sdp_attr *attr)
|
|
{
|
|
struct ast_rtp_engine_dtls *dtls = ast_rtp_instance_get_dtls(session_media->rtp);
|
|
pj_str_t *value;
|
|
|
|
if (!attr->value.ptr) {
|
|
return;
|
|
}
|
|
|
|
value = pj_strtrim(&attr->value);
|
|
|
|
if (!pj_strcmp2(&attr->name, "setup")) {
|
|
if (!pj_stricmp2(value, "active")) {
|
|
dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_ACTIVE);
|
|
} else if (!pj_stricmp2(value, "passive")) {
|
|
dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_PASSIVE);
|
|
} else if (!pj_stricmp2(value, "actpass")) {
|
|
dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_ACTPASS);
|
|
} else if (!pj_stricmp2(value, "holdconn")) {
|
|
dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_HOLDCONN);
|
|
} else {
|
|
ast_log(LOG_WARNING, "Unsupported setup attribute value '%*s'\n", (int)value->slen, value->ptr);
|
|
}
|
|
} else if (!pj_strcmp2(&attr->name, "connection")) {
|
|
if (!pj_stricmp2(value, "new")) {
|
|
dtls->reset(session_media->rtp);
|
|
} else if (!pj_stricmp2(value, "existing")) {
|
|
/* Do nothing */
|
|
} else {
|
|
ast_log(LOG_WARNING, "Unsupported connection attribute value '%*s'\n", (int)value->slen, value->ptr);
|
|
}
|
|
} else if (!pj_strcmp2(&attr->name, "fingerprint")) {
|
|
char hash_value[256], hash[32];
|
|
char fingerprint_text[value->slen + 1];
|
|
ast_copy_pj_str(fingerprint_text, value, sizeof(fingerprint_text));
|
|
if (sscanf(fingerprint_text, "%31s %255s", hash, hash_value) == 2) {
|
|
if (!strcasecmp(hash, "sha-1")) {
|
|
dtls->set_fingerprint(session_media->rtp, AST_RTP_DTLS_HASH_SHA1, hash_value);
|
|
} else if (!strcasecmp(hash, "sha-256")) {
|
|
dtls->set_fingerprint(session_media->rtp, AST_RTP_DTLS_HASH_SHA256, hash_value);
|
|
} else {
|
|
ast_log(LOG_WARNING, "Unsupported fingerprint hash type '%s'\n",
|
|
hash);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static int parse_dtls_attrib(struct ast_sip_session_media *session_media,
|
|
const struct pjmedia_sdp_session *sdp,
|
|
const struct pjmedia_sdp_media *stream)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < sdp->attr_count; i++) {
|
|
apply_dtls_attrib(session_media, sdp->attr[i]);
|
|
}
|
|
|
|
for (i = 0; i < stream->attr_count; i++) {
|
|
apply_dtls_attrib(session_media, stream->attr[i]);
|
|
}
|
|
|
|
ast_set_flag(session_media->srtp, AST_SRTP_CRYPTO_OFFER_OK);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int setup_sdes_srtp(struct ast_sip_session_media *session_media,
|
|
const struct pjmedia_sdp_media *stream)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < stream->attr_count; i++) {
|
|
pjmedia_sdp_attr *attr;
|
|
RAII_VAR(char *, crypto_str, NULL, ast_free);
|
|
|
|
/* check the stream for the required crypto attribute */
|
|
attr = stream->attr[i];
|
|
if (pj_strcmp2(&attr->name, "crypto")) {
|
|
continue;
|
|
}
|
|
|
|
crypto_str = ast_strndup(attr->value.ptr, attr->value.slen);
|
|
if (!crypto_str) {
|
|
return -1;
|
|
}
|
|
|
|
if (setup_srtp(session_media)) {
|
|
return -1;
|
|
}
|
|
|
|
if (!ast_sdp_crypto_process(session_media->rtp, session_media->srtp, crypto_str)) {
|
|
/* found a valid crypto attribute */
|
|
return 0;
|
|
}
|
|
|
|
ast_debug(1, "Ignoring crypto offer with unsupported parameters: %s\n", crypto_str);
|
|
}
|
|
|
|
/* no usable crypto attributes found */
|
|
return -1;
|
|
}
|
|
|
|
static int setup_media_encryption(struct ast_sip_session *session,
|
|
struct ast_sip_session_media *session_media,
|
|
const struct pjmedia_sdp_session *sdp,
|
|
const struct pjmedia_sdp_media *stream)
|
|
{
|
|
switch (session_media->encryption) {
|
|
case AST_SIP_MEDIA_ENCRYPT_SDES:
|
|
if (setup_sdes_srtp(session_media, stream)) {
|
|
return -1;
|
|
}
|
|
break;
|
|
case AST_SIP_MEDIA_ENCRYPT_DTLS:
|
|
if (setup_dtls_srtp(session, session_media)) {
|
|
return -1;
|
|
}
|
|
if (parse_dtls_attrib(session_media, sdp, stream)) {
|
|
return -1;
|
|
}
|
|
break;
|
|
case AST_SIP_MEDIA_TRANSPORT_INVALID:
|
|
case AST_SIP_MEDIA_ENCRYPT_NONE:
|
|
break;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function which negotiates an incoming media stream */
|
|
static int negotiate_incoming_sdp_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
|
|
const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream)
|
|
{
|
|
char host[NI_MAXHOST];
|
|
RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free_ptr);
|
|
enum ast_media_type media_type = stream_to_media_type(session_media->stream_type);
|
|
enum ast_sip_session_media_encryption encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
|
|
int res;
|
|
|
|
/* If port is 0, ignore this media stream */
|
|
if (!stream->desc.port) {
|
|
ast_debug(3, "Media stream '%s' is already declined\n", session_media->stream_type);
|
|
return 0;
|
|
}
|
|
|
|
/* If no type formats have been configured reject this stream */
|
|
if (!ast_format_cap_has_type(session->endpoint->media.codecs, media_type)) {
|
|
ast_debug(3, "Endpoint has no codecs for media type '%s', declining stream\n", session_media->stream_type);
|
|
return 0;
|
|
}
|
|
|
|
/* Ensure incoming transport is compatible with the endpoint's configuration */
|
|
if (!session->endpoint->media.rtp.use_received_transport) {
|
|
encryption = check_endpoint_media_transport(session->endpoint, stream);
|
|
|
|
if (encryption == AST_SIP_MEDIA_TRANSPORT_INVALID) {
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &sdp->conn->addr, sizeof(host));
|
|
|
|
/* Ensure that the address provided is valid */
|
|
if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
|
|
/* The provided host was actually invalid so we error out this negotiation */
|
|
return -1;
|
|
}
|
|
|
|
/* Using the connection information create an appropriate RTP instance */
|
|
if (!session_media->rtp && create_rtp(session, session_media, ast_sockaddr_is_ipv6(addrs))) {
|
|
return -1;
|
|
}
|
|
|
|
res = setup_media_encryption(session, session_media, sdp, stream);
|
|
if (res) {
|
|
if (!session->endpoint->media.rtp.encryption_optimistic) {
|
|
/* If optimistic encryption is disabled and crypto should have been enabled
|
|
* but was not this session must fail.
|
|
*/
|
|
return -1;
|
|
}
|
|
/* There is no encryption, sad. */
|
|
session_media->encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
|
|
}
|
|
|
|
/* If we've been explicitly configured to use the received transport OR if
|
|
* encryption is on and crypto is present use the received transport.
|
|
* This is done in case of optimistic because it may come in as RTP/AVP or RTP/SAVP depending
|
|
* on the configuration of the remote endpoint (optimistic themselves or mandatory).
|
|
*/
|
|
if ((session->endpoint->media.rtp.use_received_transport) ||
|
|
((encryption == AST_SIP_MEDIA_ENCRYPT_SDES) && !res)) {
|
|
pj_strdup(session->inv_session->pool, &session_media->transport, &stream->desc.transport);
|
|
}
|
|
|
|
if (set_caps(session, session_media, stream)) {
|
|
return 0;
|
|
}
|
|
return 1;
|
|
}
|
|
|
|
static int add_crypto_to_stream(struct ast_sip_session *session,
|
|
struct ast_sip_session_media *session_media,
|
|
pj_pool_t *pool, pjmedia_sdp_media *media)
|
|
{
|
|
pj_str_t stmp;
|
|
pjmedia_sdp_attr *attr;
|
|
enum ast_rtp_dtls_hash hash;
|
|
const char *crypto_attribute;
|
|
struct ast_rtp_engine_dtls *dtls;
|
|
static const pj_str_t STR_NEW = { "new", 3 };
|
|
static const pj_str_t STR_EXISTING = { "existing", 8 };
|
|
static const pj_str_t STR_ACTIVE = { "active", 6 };
|
|
static const pj_str_t STR_PASSIVE = { "passive", 7 };
|
|
static const pj_str_t STR_ACTPASS = { "actpass", 7 };
|
|
static const pj_str_t STR_HOLDCONN = { "holdconn", 8 };
|
|
|
|
switch (session_media->encryption) {
|
|
case AST_SIP_MEDIA_ENCRYPT_NONE:
|
|
case AST_SIP_MEDIA_TRANSPORT_INVALID:
|
|
break;
|
|
case AST_SIP_MEDIA_ENCRYPT_SDES:
|
|
if (!session_media->srtp) {
|
|
session_media->srtp = ast_sdp_srtp_alloc();
|
|
if (!session_media->srtp) {
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
crypto_attribute = ast_sdp_srtp_get_attrib(session_media->srtp,
|
|
0 /* DTLS running? No */,
|
|
session->endpoint->media.rtp.srtp_tag_32 /* 32 byte tag length? */);
|
|
if (!crypto_attribute) {
|
|
/* No crypto attribute to add, bad news */
|
|
return -1;
|
|
}
|
|
|
|
attr = pjmedia_sdp_attr_create(pool, "crypto", pj_cstr(&stmp, crypto_attribute));
|
|
media->attr[media->attr_count++] = attr;
|
|
break;
|
|
case AST_SIP_MEDIA_ENCRYPT_DTLS:
|
|
if (setup_dtls_srtp(session, session_media)) {
|
|
return -1;
|
|
}
|
|
|
|
dtls = ast_rtp_instance_get_dtls(session_media->rtp);
|
|
if (!dtls) {
|
|
return -1;
|
|
}
|
|
|
|
switch (dtls->get_connection(session_media->rtp)) {
|
|
case AST_RTP_DTLS_CONNECTION_NEW:
|
|
attr = pjmedia_sdp_attr_create(pool, "connection", &STR_NEW);
|
|
media->attr[media->attr_count++] = attr;
|
|
break;
|
|
case AST_RTP_DTLS_CONNECTION_EXISTING:
|
|
attr = pjmedia_sdp_attr_create(pool, "connection", &STR_EXISTING);
|
|
media->attr[media->attr_count++] = attr;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
switch (dtls->get_setup(session_media->rtp)) {
|
|
case AST_RTP_DTLS_SETUP_ACTIVE:
|
|
attr = pjmedia_sdp_attr_create(pool, "setup", &STR_ACTIVE);
|
|
media->attr[media->attr_count++] = attr;
|
|
break;
|
|
case AST_RTP_DTLS_SETUP_PASSIVE:
|
|
attr = pjmedia_sdp_attr_create(pool, "setup", &STR_PASSIVE);
|
|
media->attr[media->attr_count++] = attr;
|
|
break;
|
|
case AST_RTP_DTLS_SETUP_ACTPASS:
|
|
attr = pjmedia_sdp_attr_create(pool, "setup", &STR_ACTPASS);
|
|
media->attr[media->attr_count++] = attr;
|
|
break;
|
|
case AST_RTP_DTLS_SETUP_HOLDCONN:
|
|
attr = pjmedia_sdp_attr_create(pool, "setup", &STR_HOLDCONN);
|
|
media->attr[media->attr_count++] = attr;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
hash = dtls->get_fingerprint_hash(session_media->rtp);
|
|
crypto_attribute = dtls->get_fingerprint(session_media->rtp);
|
|
if (crypto_attribute && (hash == AST_RTP_DTLS_HASH_SHA1 || hash == AST_RTP_DTLS_HASH_SHA256)) {
|
|
RAII_VAR(struct ast_str *, fingerprint, ast_str_create(64), ast_free);
|
|
if (!fingerprint) {
|
|
return -1;
|
|
}
|
|
|
|
if (hash == AST_RTP_DTLS_HASH_SHA1) {
|
|
ast_str_set(&fingerprint, 0, "SHA-1 %s", crypto_attribute);
|
|
} else {
|
|
ast_str_set(&fingerprint, 0, "SHA-256 %s", crypto_attribute);
|
|
}
|
|
|
|
attr = pjmedia_sdp_attr_create(pool, "fingerprint", pj_cstr(&stmp, ast_str_buffer(fingerprint)));
|
|
media->attr[media->attr_count++] = attr;
|
|
}
|
|
break;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function which creates an outgoing stream */
|
|
static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
|
|
struct pjmedia_sdp_session *sdp)
|
|
{
|
|
pj_pool_t *pool = session->inv_session->pool_prov;
|
|
static const pj_str_t STR_IN = { "IN", 2 };
|
|
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;
|
|
char tmp[512];
|
|
pj_str_t stmp;
|
|
pjmedia_sdp_attr *attr;
|
|
int index = 0;
|
|
int noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) ? AST_RTP_DTMF : 0;
|
|
int min_packet_size = 0, max_packet_size = 0;
|
|
int rtp_code;
|
|
RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
|
|
enum ast_media_type media_type = stream_to_media_type(session_media->stream_type);
|
|
int use_override_prefs = ast_format_cap_count(session->req_caps);
|
|
|
|
int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
|
|
ast_format_cap_count(session->direct_media_cap);
|
|
|
|
if ((use_override_prefs && !ast_format_cap_has_type(session->req_caps, media_type)) ||
|
|
(!use_override_prefs && !ast_format_cap_has_type(session->endpoint->media.codecs, media_type))) {
|
|
/* If no type formats are configured don't add a stream */
|
|
return 0;
|
|
} else if (!session_media->rtp && create_rtp(session, session_media, session->endpoint->media.rtp.ipv6)) {
|
|
return -1;
|
|
}
|
|
|
|
if (!(media = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_media))) ||
|
|
!(media->conn = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_conn)))) {
|
|
return -1;
|
|
}
|
|
|
|
if (add_crypto_to_stream(session, session_media, pool, media)) {
|
|
return -1;
|
|
}
|
|
|
|
media->desc.media = pj_str(session_media->stream_type);
|
|
if (pj_strlen(&session_media->transport)) {
|
|
/* If a transport has already been specified use it */
|
|
media->desc.transport = session_media->transport;
|
|
} else {
|
|
media->desc.transport = pj_str(ast_sdp_get_rtp_profile(
|
|
/* Optimistic encryption places crypto in the normal RTP/AVP profile */
|
|
!session->endpoint->media.rtp.encryption_optimistic &&
|
|
(session_media->encryption == AST_SIP_MEDIA_ENCRYPT_SDES),
|
|
session_media->rtp, session->endpoint->media.rtp.use_avpf,
|
|
session->endpoint->media.rtp.force_avp));
|
|
}
|
|
|
|
/* Add connection level details */
|
|
if (direct_media_enabled) {
|
|
ast_copy_string(hostip, ast_sockaddr_stringify_fmt(&session_media->direct_media_addr, AST_SOCKADDR_STR_ADDR), sizeof(hostip));
|
|
} else if (ast_strlen_zero(session->endpoint->media.address)) {
|
|
pj_sockaddr localaddr;
|
|
|
|
if (pj_gethostip(session->endpoint->media.rtp.ipv6 ? pj_AF_INET6() : pj_AF_INET(), &localaddr)) {
|
|
return -1;
|
|
}
|
|
pj_sockaddr_print(&localaddr, hostip, sizeof(hostip), 2);
|
|
} else {
|
|
ast_copy_string(hostip, session->endpoint->media.address, sizeof(hostip));
|
|
}
|
|
|
|
media->conn->net_type = STR_IN;
|
|
media->conn->addr_type = session->endpoint->media.rtp.ipv6 ? STR_IP6 : STR_IP4;
|
|
pj_strdup2(pool, &media->conn->addr, hostip);
|
|
ast_rtp_instance_get_local_address(session_media->rtp, &addr);
|
|
media->desc.port = direct_media_enabled ? ast_sockaddr_port(&session_media->direct_media_addr) : (pj_uint16_t) ast_sockaddr_port(&addr);
|
|
media->desc.port_count = 1;
|
|
|
|
/* Add ICE attributes and candidates */
|
|
add_ice_to_stream(session, session_media, pool, media);
|
|
|
|
if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
|
|
ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n", session_media->stream_type);
|
|
return -1;
|
|
}
|
|
|
|
if (direct_media_enabled) {
|
|
ast_format_cap_get_compatible(session->endpoint->media.codecs, session->direct_media_cap, caps);
|
|
} else if (!ast_format_cap_count(session->req_caps) ||
|
|
!ast_format_cap_iscompatible(session->req_caps, session->endpoint->media.codecs)) {
|
|
ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, media_type);
|
|
} else {
|
|
ast_format_cap_append_from_cap(caps, session->req_caps, media_type);
|
|
}
|
|
|
|
for (index = 0; index < ast_format_cap_count(caps); ++index) {
|
|
struct ast_format *format = ast_format_cap_get_format(caps, index);
|
|
|
|
if (ast_format_get_type(format) != media_type) {
|
|
ao2_ref(format, -1);
|
|
continue;
|
|
}
|
|
|
|
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0)) == -1) {
|
|
ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format));
|
|
ao2_ref(format, -1);
|
|
continue;
|
|
}
|
|
|
|
if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 1, format, 0))) {
|
|
ao2_ref(format, -1);
|
|
continue;
|
|
}
|
|
media->attr[media->attr_count++] = attr;
|
|
|
|
if ((attr = generate_fmtp_attr(pool, format, rtp_code))) {
|
|
media->attr[media->attr_count++] = attr;
|
|
}
|
|
|
|
if (ast_format_get_maximum_ms(format) &&
|
|
((ast_format_get_maximum_ms(format) < max_packet_size) || !max_packet_size)) {
|
|
max_packet_size = ast_format_get_maximum_ms(format);
|
|
}
|
|
ao2_ref(format, -1);
|
|
}
|
|
|
|
/* Add non-codec formats */
|
|
if (media_type != AST_MEDIA_TYPE_VIDEO) {
|
|
for (index = 1LL; index <= AST_RTP_MAX; index <<= 1) {
|
|
if (!(noncodec & index) || (rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp),
|
|
0, NULL, index)) == -1) {
|
|
continue;
|
|
}
|
|
|
|
if (!(attr = generate_rtpmap_attr(media, pool, rtp_code, 0, NULL, index))) {
|
|
continue;
|
|
}
|
|
|
|
media->attr[media->attr_count++] = attr;
|
|
|
|
if (index == AST_RTP_DTMF) {
|
|
snprintf(tmp, sizeof(tmp), "%d 0-16", rtp_code);
|
|
attr = pjmedia_sdp_attr_create(pool, "fmtp", pj_cstr(&stmp, tmp));
|
|
media->attr[media->attr_count++] = attr;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* If no formats were actually added to the media stream don't add it to the SDP */
|
|
if (!media->desc.fmt_count) {
|
|
return 1;
|
|
}
|
|
|
|
/* If ptime is set add it as an attribute */
|
|
min_packet_size = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(session_media->rtp));
|
|
if (!min_packet_size) {
|
|
min_packet_size = ast_format_cap_get_framing(caps);
|
|
}
|
|
if (min_packet_size) {
|
|
snprintf(tmp, sizeof(tmp), "%d", min_packet_size);
|
|
attr = pjmedia_sdp_attr_create(pool, "ptime", pj_cstr(&stmp, tmp));
|
|
media->attr[media->attr_count++] = attr;
|
|
}
|
|
|
|
if (max_packet_size) {
|
|
snprintf(tmp, sizeof(tmp), "%d", max_packet_size);
|
|
attr = pjmedia_sdp_attr_create(pool, "maxptime", pj_cstr(&stmp, tmp));
|
|
media->attr[media->attr_count++] = attr;
|
|
}
|
|
|
|
/* 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 = !session_media->locally_held ? STR_SENDRECV : STR_SENDONLY;
|
|
media->attr[media->attr_count++] = attr;
|
|
|
|
/* Add the media stream to the SDP */
|
|
sdp->media[sdp->media_count++] = media;
|
|
|
|
return 1;
|
|
}
|
|
|
|
static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
|
|
const struct pjmedia_sdp_session *local, const struct pjmedia_sdp_media *local_stream,
|
|
const struct pjmedia_sdp_session *remote, const struct pjmedia_sdp_media *remote_stream)
|
|
{
|
|
RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free_ptr);
|
|
enum ast_media_type media_type = stream_to_media_type(session_media->stream_type);
|
|
char host[NI_MAXHOST];
|
|
int fdno, res;
|
|
|
|
if (!session->channel) {
|
|
return 1;
|
|
}
|
|
|
|
if (!local_stream->desc.port || !remote_stream->desc.port) {
|
|
return 1;
|
|
}
|
|
|
|
/* Ensure incoming transport is compatible with the endpoint's configuration */
|
|
if (!session->endpoint->media.rtp.use_received_transport &&
|
|
check_endpoint_media_transport(session->endpoint, remote_stream) == AST_SIP_MEDIA_TRANSPORT_INVALID) {
|
|
return -1;
|
|
}
|
|
|
|
/* Create an RTP instance if need be */
|
|
if (!session_media->rtp && create_rtp(session, session_media, session->endpoint->media.rtp.ipv6)) {
|
|
return -1;
|
|
}
|
|
|
|
res = setup_media_encryption(session, session_media, remote, remote_stream);
|
|
if (!session->endpoint->media.rtp.encryption_optimistic && res) {
|
|
/* If optimistic encryption is disabled and crypto should have been enabled but was not
|
|
* this session must fail.
|
|
*/
|
|
return -1;
|
|
}
|
|
|
|
if (!remote_stream->conn && !remote->conn) {
|
|
return 1;
|
|
}
|
|
|
|
ast_copy_pj_str(host, remote_stream->conn ? &remote_stream->conn->addr : &remote->conn->addr, sizeof(host));
|
|
|
|
/* Ensure that the address provided is valid */
|
|
if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
|
|
/* The provided host was actually invalid so we error out this negotiation */
|
|
return -1;
|
|
}
|
|
|
|
/* Apply connection information to the RTP instance */
|
|
ast_sockaddr_set_port(addrs, remote_stream->desc.port);
|
|
ast_rtp_instance_set_remote_address(session_media->rtp, addrs);
|
|
if (set_caps(session, session_media, local_stream)) {
|
|
return 1;
|
|
}
|
|
|
|
if ((fdno = media_type_to_fdno(media_type)) < 0) {
|
|
return -1;
|
|
}
|
|
ast_channel_set_fd(session->channel, fdno, ast_rtp_instance_fd(session_media->rtp, 0));
|
|
ast_channel_set_fd(session->channel, fdno + 1, ast_rtp_instance_fd(session_media->rtp, 1));
|
|
|
|
/* If ICE support is enabled find all the needed attributes */
|
|
process_ice_attributes(session, session_media, remote, remote_stream);
|
|
|
|
/* Ensure the RTP instance is active */
|
|
ast_rtp_instance_activate(session_media->rtp);
|
|
|
|
/* audio stream handles music on hold */
|
|
if (media_type != AST_MEDIA_TYPE_AUDIO) {
|
|
if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE)
|
|
&& (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) {
|
|
ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER);
|
|
}
|
|
return 1;
|
|
}
|
|
|
|
if (ast_sockaddr_isnull(addrs) ||
|
|
ast_sockaddr_is_any(addrs) ||
|
|
pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL) ||
|
|
pjmedia_sdp_media_find_attr2(remote_stream, "inactive", NULL)) {
|
|
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->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->remotely_held = 0;
|
|
} else if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE)
|
|
&& (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) {
|
|
ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER);
|
|
}
|
|
|
|
/* This purposely resets the encryption to the configured in case it gets added later */
|
|
session_media->encryption = session->endpoint->media.rtp.encryption;
|
|
|
|
return 1;
|
|
}
|
|
|
|
/*! \brief Function which updates the media stream with external media address, if applicable */
|
|
static void change_outgoing_sdp_stream_media_address(pjsip_tx_data *tdata, struct pjmedia_sdp_media *stream, struct ast_sip_transport *transport)
|
|
{
|
|
char host[NI_MAXHOST];
|
|
struct ast_sockaddr addr = { { 0, } };
|
|
|
|
/* If the stream has been rejected there will be no connection line */
|
|
if (!stream->conn) {
|
|
return;
|
|
}
|
|
|
|
ast_copy_pj_str(host, &stream->conn->addr, sizeof(host));
|
|
ast_sockaddr_parse(&addr, host, PARSE_PORT_FORBID);
|
|
|
|
/* Is the address within the SDP inside the same network? */
|
|
if (ast_apply_ha(transport->localnet, &addr) == AST_SENSE_ALLOW) {
|
|
return;
|
|
}
|
|
|
|
pj_strdup2(tdata->pool, &stream->conn->addr, transport->external_media_address);
|
|
}
|
|
|
|
/*! \brief Function which destroys the RTP instance when session ends */
|
|
static void stream_destroy(struct ast_sip_session_media *session_media)
|
|
{
|
|
if (session_media->rtp) {
|
|
ast_rtp_instance_stop(session_media->rtp);
|
|
ast_rtp_instance_destroy(session_media->rtp);
|
|
}
|
|
session_media->rtp = NULL;
|
|
}
|
|
|
|
/*! \brief SDP handler for 'audio' media stream */
|
|
static struct ast_sip_session_sdp_handler audio_sdp_handler = {
|
|
.id = STR_AUDIO,
|
|
.negotiate_incoming_sdp_stream = negotiate_incoming_sdp_stream,
|
|
.create_outgoing_sdp_stream = create_outgoing_sdp_stream,
|
|
.apply_negotiated_sdp_stream = apply_negotiated_sdp_stream,
|
|
.change_outgoing_sdp_stream_media_address = change_outgoing_sdp_stream_media_address,
|
|
.stream_destroy = stream_destroy,
|
|
};
|
|
|
|
/*! \brief SDP handler for 'video' media stream */
|
|
static struct ast_sip_session_sdp_handler video_sdp_handler = {
|
|
.id = STR_VIDEO,
|
|
.negotiate_incoming_sdp_stream = negotiate_incoming_sdp_stream,
|
|
.create_outgoing_sdp_stream = create_outgoing_sdp_stream,
|
|
.apply_negotiated_sdp_stream = apply_negotiated_sdp_stream,
|
|
.change_outgoing_sdp_stream_media_address = change_outgoing_sdp_stream_media_address,
|
|
.stream_destroy = stream_destroy,
|
|
};
|
|
|
|
static int video_info_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
|
|
{
|
|
struct pjsip_transaction *tsx;
|
|
pjsip_tx_data *tdata;
|
|
|
|
if (!session->channel
|
|
|| !ast_sip_is_content_type(&rdata->msg_info.msg->body->content_type,
|
|
"application",
|
|
"media_control+xml")) {
|
|
return 0;
|
|
}
|
|
|
|
tsx = pjsip_rdata_get_tsx(rdata);
|
|
|
|
ast_queue_control(session->channel, AST_CONTROL_VIDUPDATE);
|
|
|
|
if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, 200, NULL, &tdata) == PJ_SUCCESS) {
|
|
pjsip_dlg_send_response(session->inv_session->dlg, tsx, tdata);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static struct ast_sip_session_supplement video_info_supplement = {
|
|
.method = "INFO",
|
|
.incoming_request = video_info_incoming_request,
|
|
};
|
|
|
|
/*! \brief Unloads the sdp RTP/AVP module from Asterisk */
|
|
static int unload_module(void)
|
|
{
|
|
ast_sip_session_unregister_supplement(&video_info_supplement);
|
|
ast_sip_session_unregister_sdp_handler(&video_sdp_handler, STR_VIDEO);
|
|
ast_sip_session_unregister_sdp_handler(&audio_sdp_handler, STR_AUDIO);
|
|
|
|
if (sched) {
|
|
ast_sched_context_destroy(sched);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*!
|
|
* \brief Load the module
|
|
*
|
|
* Module loading including tests for configuration or dependencies.
|
|
* This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
|
|
* or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
|
|
* tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
|
|
* configuration file or other non-critical problem return
|
|
* AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
|
|
*/
|
|
static int load_module(void)
|
|
{
|
|
CHECK_PJSIP_SESSION_MODULE_LOADED();
|
|
|
|
ast_sockaddr_parse(&address_ipv4, "0.0.0.0", 0);
|
|
ast_sockaddr_parse(&address_ipv6, "::", 0);
|
|
|
|
if (!(sched = ast_sched_context_create())) {
|
|
ast_log(LOG_ERROR, "Unable to create scheduler context.\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_sched_start_thread(sched)) {
|
|
ast_log(LOG_ERROR, "Unable to create scheduler context thread.\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_sip_session_register_sdp_handler(&audio_sdp_handler, STR_AUDIO)) {
|
|
ast_log(LOG_ERROR, "Unable to register SDP handler for %s stream type\n", STR_AUDIO);
|
|
goto end;
|
|
}
|
|
|
|
if (ast_sip_session_register_sdp_handler(&video_sdp_handler, STR_VIDEO)) {
|
|
ast_log(LOG_ERROR, "Unable to register SDP handler for %s stream type\n", STR_VIDEO);
|
|
goto end;
|
|
}
|
|
|
|
ast_sip_session_register_supplement(&video_info_supplement);
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
end:
|
|
unload_module();
|
|
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP SDP RTP/AVP stream handler",
|
|
.support_level = AST_MODULE_SUPPORT_CORE,
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
|
|
);
|