Freeswitch支持从传入的3PCC INVITE生成SRTP提议？

Question

我在Node.js中使用Freeswitch ESL（事件套接字库）当我从一个经典的sip端点接收到SIP.js的邀请时，SDP内容是常规的RTP.我需要能够在中间使用Freeswitch将RTP端点连接到SRTP端点，为了做到这一点，我们需要能够在Freeswitch上创建一个没有SDP的端点（3PCC INVITE），但我们希望这个端点能够返回SRTP SDP而不是RTP SDP。

主要的问题是：Freeswitch是否支持生成SRTP SDP？Freeswitch是否支持从传入的3PCC INVITE生成SRTP offer？

我相信这是Freeswitch 1.8.5中处理3PCC INVITE的部分。https:/github.comsignalwirefreeswitchblobv1.8.5srcmodendpointsmod_sofiasofia.c。

if (sofia_test_pflag(profile, PFLAG_3PCC)) {
    if (switch_channel_test_flag(channel, CF_PROXY_MODE) || switch_channel_test_flag(channel, CF_PROXY_MEDIA)) {
        switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "No SDP in INVITE and 3pcc=yes cannot work with bypass or proxy media, hanging up.\n");
        switch_channel_set_variable(channel, SWITCH_ENDPOINT_DISPOSITION_VARIABLE, "3PCC DISABLED");
        switch_channel_hangup(channel, SWITCH_CAUSE_MANDATORY_IE_MISSING);
    } else {
        switch_channel_set_variable(channel, SWITCH_ENDPOINT_DISPOSITION_VARIABLE, "RECEIVED_NOSDP");
        switch_core_media_choose_port(tech_pvt->session, SWITCH_MEDIA_TYPE_AUDIO, 0);
        switch_core_media_prepare_codecs(session, 1);
        switch_channel_set_state(channel, CS_HIBERNATE);
        switch_core_media_gen_local_sdp(session, SDP_TYPE_REQUEST, NULL, 0, NULL, 0);
        sofia_set_flag_locked(tech_pvt, TFLAG_3PCC);

        if (sofia_use_soa(tech_pvt)) {
            nua_respond(tech_pvt->nh, SIP_200_OK,
                        SIPTAG_CONTACT_STR(tech_pvt->reply_contact),
                        SOATAG_USER_SDP_STR(tech_pvt->mparams.local_sdp_str),
                        SOATAG_REUSE_REJECTED(1),
                        SOATAG_AUDIO_AUX("cn telephone-event"),
                        TAG_IF(sofia_test_pflag(profile, PFLAG_DISABLE_100REL), NUTAG_INCLUDE_EXTRA_SDP(1)), TAG_END());
        } else {
            nua_respond(tech_pvt->nh, SIP_200_OK,
                        NUTAG_MEDIA_ENABLE(0),
                        SIPTAG_CONTACT_STR(tech_pvt->reply_contact),
                        SIPTAG_CONTENT_TYPE_STR("application/sdp"), SIPTAG_PAYLOAD_STR(tech_pvt->mparams.local_sdp_str), TAG_END());
        }
    }
}

也许可以使用他们用于mod_verto的相同模式。https:/github.comsignalwirefreeswitchblobv1.8.5srcmodendpointsmod_vertomod_verto.c。

这个函数 switch_core_media_gen_local_sdp. 似乎是创建RTP SDP的那个。https:/github.comsignalwirefreeswitchblobv1.8.5srcswitch_core_media.c。如何改成SRTP SDP呢？

static const char *get_media_profile_name(switch_core_session_t *session, int secure)
{
    switch_assert(session);

    if (switch_channel_test_flag(session->channel, CF_AVPF)) {
        if (switch_channel_test_flag(session->channel, CF_DTLS) || secure) {
            if (switch_channel_test_flag(session->channel, CF_AVPF_MOZ)) {
                return "UDP/TLS/RTP/SAVPF";
            } else {
                return "RTP/SAVPF";
            }
        } else {
            if (switch_channel_test_flag(session->channel, CF_AVPF_MOZ)) {
                return "UDP/AVPF";
            } else {
                return "RTP/AVPF";
            }
        }
    }

    if (secure) {
        return "RTP/SAVP";
    }

    return "RTP/AVP";

}

启用SRTP有3个条件。

session->channel->flags['CF_AVPF']
(session->channel->flags['CF_DTLS'] || secure)
session->channel->flags['CF_AVPF_MOZ']

要把条件 CF_AVPF 旗帜鲜明 开关_核心_会话_设置_冰块 或。

if (is_outbound || switch_channel_test_flag(session->channel, CF_RECOVERING) ||
    switch_channel_test_flag(session->channel, CF_3PCC)) {
    if (!switch_channel_test_flag(session->channel, CF_AVPF) &&
        switch_true(switch_channel_get_variable(session->channel, "media_webrtc"))) {
        switch_channel_set_flag(session->channel, CF_AVPF);
        switch_channel_set_flag(session->channel, CF_ICE);
        smh->mparams->rtcp_audio_interval_msec = SWITCH_RTCP_AUDIO_INTERVAL_MSEC;
        smh->mparams->rtcp_video_interval_msec = SWITCH_RTCP_VIDEO_INTERVAL_MSEC;
    }

    if (switch_true(switch_channel_get_variable(session->channel, "add_ice_candidates"))) {
        switch_channel_set_flag(session->channel, CF_ICE);
    }

    if ( switch_rtp_has_dtls() && dtls_ok(session)) {
        if (switch_channel_test_flag(session->channel, CF_AVPF) ||
            switch_true(switch_channel_get_variable(smh->session->channel, "rtp_use_dtls"))) {
            switch_channel_set_flag(smh->session->channel, CF_DTLS);
            switch_channel_set_flag(smh->session->channel, CF_SECURE);
            generate_local_fingerprint(smh, SWITCH_MEDIA_TYPE_AUDIO);
        }
    }
    switch_core_session_parse_crypto_prefs(session);
    switch_core_session_check_outgoing_crypto(session);
}

落实条件 CF_DTLS 标志与上述相同(CF_AVPF)或调用 检查_冰块:

if (switch_rtp_has_dtls() && dtls_ok(smh->session) && !strcasecmp(attr->a_name, "fingerprint") && !zstr(attr->a_value)) {
    char *p;

    engine->remote_dtls_fingerprint.type = switch_core_session_strdup(smh->session, attr->a_value);

    if ((p = strchr(engine->remote_dtls_fingerprint.type, ' '))) {
        *p++ = '\0';

        if (switch_channel_test_flag(smh->session->channel, CF_REINVITE) && !switch_channel_test_flag(smh->session->channel, CF_RECOVERING) &&
            !zstr(engine->remote_dtls_fingerprint.str) && !strcmp(engine->remote_dtls_fingerprint.str, p)) {
            engine->new_dtls = 0;
        } else {
            switch_set_string(engine->remote_dtls_fingerprint.str, p);
            engine->new_dtls = 1;
            engine->new_ice = 1;
        }
    }

    generate_local_fingerprint(smh, type);
    switch_channel_set_flag(smh->session->channel, CF_DTLS);

}

也可以将 "true "发送到 "true"。确保 参数，但条件很多。

将条件放在 CF_AVPF_MOZ 标志。

if (m->m_proto_name && !strcasecmp(m->m_proto_name, "UDP/TLS/RTP/SAVPF")) {
    switch_channel_set_flag(session->channel, CF_AVPF_MOZ);
}

if (m->m_proto_name && !strcasecmp(m->m_proto_name, "UDP/RTP/AVPF")) {
    switch_channel_set_flag(session->channel, CF_AVPF_MOZ);
}

Answer 1

试着设置media_webrtc=true，这应该会告诉freeswitch使用DTLS作为媒体流，WebRTC信令应该单独完成。

<action application="bridge" data="[media_webrtc=true]sofia/internal/100%${sip_profile}"/>