diff options
Diffstat (limited to 'res/res_pjsip/pjsip_configuration.c')
| -rw-r--r-- | res/res_pjsip/pjsip_configuration.c | 27 |
1 files changed, 25 insertions, 2 deletions
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c index c60173721..9f9de36fa 100644 --- a/res/res_pjsip/pjsip_configuration.c +++ b/res/res_pjsip/pjsip_configuration.c @@ -1363,8 +1363,30 @@ static int sip_endpoint_apply_handler(const struct ast_sorcery *sorcery, void *o return -1; } - if (endpoint->media.bundle) { - endpoint->media.rtcp_mux = 1; + endpoint->media.rtcp_mux |= endpoint->media.bundle; + + /* + * If webrtc has been enabled then enable those attributes, and default + * some, that are needed in order for webrtc to work. + */ + endpoint->media.bundle |= endpoint->media.webrtc; + endpoint->media.rtcp_mux |= endpoint->media.webrtc; + endpoint->media.rtp.use_avpf |= endpoint->media.webrtc; + endpoint->media.rtp.ice_support |= endpoint->media.webrtc; + endpoint->media.rtp.use_received_transport |= endpoint->media.webrtc; + + if (endpoint->media.webrtc) { + endpoint->media.rtp.encryption = AST_SIP_MEDIA_ENCRYPT_DTLS; + endpoint->media.rtp.dtls_cfg.enabled = 1; + endpoint->media.rtp.dtls_cfg.default_setup = AST_RTP_DTLS_SETUP_ACTPASS; + endpoint->media.rtp.dtls_cfg.verify = AST_RTP_DTLS_VERIFY_FINGERPRINT; + + if (ast_strlen_zero(endpoint->media.rtp.dtls_cfg.certfile) || + (ast_strlen_zero(endpoint->media.rtp.dtls_cfg.cafile))) { + ast_log(LOG_ERROR, "WebRTC can't be enabled on endpoint '%s' - a DTLS cert " + "or ca file has not been specified", ast_sorcery_object_get_id(endpoint)); + return -1; + } } return 0; @@ -1990,6 +2012,7 @@ int ast_res_pjsip_initialize_configuration(void) ast_sorcery_object_field_register(sip_sorcery, "endpoint", "max_audio_streams", "1", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.max_audio_streams)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "max_video_streams", "1", OPT_UINT_T, 0, FLDSET(struct ast_sip_endpoint, media.max_video_streams)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "bundle", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.bundle)); + ast_sorcery_object_field_register(sip_sorcery, "endpoint", "webrtc", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.webrtc)); if (ast_sip_initialize_sorcery_transport()) { ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n"); |