Hi, I tried with gstreamer implementation, the messaging works fine. But the audio call still had some problems with RTP connection I think . It looks like the problems are from ssc_media_gst.c line 726 priv_update_tx_elements(SscMediaGst *self) . So sdp_media_connections(m) returns nil, anyone knows any clue?
regards Cheng Here is the debug print: =============== ofsip_cli[1558]: GLIB MESSAGE default - This program is linked against GStreamer 0.10.18 su_source_port_create() returns 0x78fd8 sofsip_cli[1558]: GLIB DEBUG default - priv_verify_required_elements:191 sofsip_cli[1558]: GLIB MESSAGE default - Verifying GST element "mulawenc" -> OK sofsip_cli[1558]: GLIB MESSAGE default - Verifying GST element "mulawdec" -> OK sofsip_cli[1558]: GLIB MESSAGE default - Verifying GST element "udpsink" -> OK sofsip_cli[1558]: GLIB MESSAGE default - Verifying GST element "udpsrc" -> OK sofsip_cli[1558]: GLIB DEBUG default - ssc_media_class_init:124 sofsip_cli[1558]: GLIB DEBUG default - ssc_media_gst_class_init:138 sofsip_cli[1558]: GLIB DEBUG default - ssc_media_init:169 sofsip_cli[1558]: GLIB MESSAGE default - Selecting media implementation: gstreamer sofsip_cli[1558]: GLIB DEBUG default - priv_static_capabilities_gst sofsip> UA: unknown event 'nua_r_set_params' (23): 200 OK ::tag_null: 0 NOTE: destroying handle (nil). sofsip> UA: nua_r_getparams: 200 OK sip::from: sip:192.168.1.98 sip::from_str: "sip:192.168.1.98" nua::retry_count: 3 nua::max_subscriptions: 20 nua::media_enable: true nua::enableInvite: true nua::autoAlert: true nua::early_media: false nua::only183_100rel: false nua::autoAnswer: false nua::autoACK: true nua::invite_timer: 120 nua::session_timer: 0 nua::min_se: 120 nua::session_refresher: 0 nua::update_refresh: false nua::enableMessage: true nua::enableMessenger: false nua::callee_caps: false nua::media_features: false nua::service_route_enable: true nua::path_enable: true nua::refer_expires: 300 nua::refer_with_id: true nua::substate: 2 nua::substate: 3600 sip::supported: timer, 100rel sip::supported_str: "timer, 100rel" sip::allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE sip::allow_str: "INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE" nua::appl_method: "INVITE, REGISTER, PUBLISH, SUBSCRIBE" sip::user_agent: sofia-sip/1.12.10 sip::user_agent_str: "sofia-sip/1.12.10" nua::user_agent: "sofia-sip/1.12.10" nua::keepalive: 120000 nua::outbound: "natify" nta::contact: <sip:192.168.1.98> nta::udp_mtu: 1300 nta::max_proceeding: 4294967295 nta::sip_t1: 500 nta::sip_t2: 4000 nta::sip_t4: 5000 nta::sip_t1x64: 32000 nta::debug_drop_prob: 0 nta::default_proxy: <null> nta::aliases: <NONE> nta::sipflags: 2 soa::caps_sdp: v=0 o=- 5257949740770292933 7676228063588076813 IN IP4 192.168.1.98 s=- c=IN IP4 192.168.1.98 t=0 0 m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 soa::caps_sdp_str: "v=0 o=- 5257949740770292933 7676228063588076813 IN IP4 192.168.1.98 s=- c=IN IP4 192.168.1.98 t=0 0 m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 " soa::user_sdp: v=0 m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 soa::user_sdp_str: "v=0 m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 " soa::local_sdp_str: <null> soa::af: 3 soa::srtp_enable: false soa::srtp_confidentiality: false soa::srtp_integrity: false ::tag_null: 0 Starting sofsip-cli in interactive mode. Issue 'h' to get list of available commands. sofsip> i sip:192.168.1.97 sofsip_cli[1558]: GLIB DEBUG default - priv_activate_gst sofsip_cli[1558]: GLIB DEBUG default - priv_static_capabilities_gst Set local SDP based on capabilities: v=0 m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 sofsip_cli[1558]: GLIB DEBUG default - priv_set_local_sdp sofsip_cli[1558]: GLIB DEBUG default - priv_update_tx_elements sofsip_cli[1558]: GLIB MESSAGE default - No RTP destination available (r_sdp=(nil)).Set remote SDP based on capabilities: v=0 m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 sofsip_cli[1558]: GLIB DEBUG default - priv_set_remote_sdp sofsip_cli[1558]: GLIB DEBUG default - priv_update_tx_elements sofsip_cli[1558]: GLIB MESSAGE default - No RTP destination available (r_sdp=0x85a6c). sofsip_cli[1558]: GLIB DEBUG default - priv_activate_gst sofsip_cli[1558]: GLIB DEBUG default - priv_setup_rtpelements Succesfully bound to local port 16384. su_source_port_create() returns 0x86690 sofsip_cli[1558]: GLIB DEBUG default - Setting sockfd to 10. sofsip_cli[1558]: GLIB DEBUG default - Setting aux-sockfd to -1. sofsip_cli[1558]: GLIB DEBUG default - farsight_netsocket_stun_map sofsip_cli[1558]: GLIB DEBUG default - server:(null) domain:192.168.1.98. UA: INVITE to sip:192.168.1.97 pending sofsip> sofsip_cli[1558]: GLIB DEBUG default - cb_stun_state: stun_error (8) sofsip_cli[1558]: GLIB DEBUG default - cb_stun_state: error in communication with the STUN server sofsip_cli[1558]: GLIB DEBUG default - priv_cb_ready sofsip_cli[1558]: GLIB MESSAGE default - Created audio sink of type 'ALSA' for playback. sofsip_cli[1558]: GLIB DEBUG default - priv_update_rx_elements sofsip_cli[1558]: GLIB DEBUG default - priv_update_rx_elements: not modifying local contact address sofsip_cli[1558]: GLIB DEBUG default - priv_update_tx_elements sofsip_cli[1558]: GLIB MESSAGE default - No RTP destination available (r_sdp=0x85a6c). sofsip_cli[1558]: GLIB MESSAGE default - Starting the pipeline. Signaling media subsystem change from 0 to 2. sofsip_cli[1558]: GLIB DEBUG default - priv_media_state_cb, state 2 UA: about to make a call with local SDP: v=0 m=audio 16384 RTP/AVP 0 a=rtpmap:0 PCMU/8000 UA: INVITE to sip:192.168.1.97 sofsip_cli[1558]: GLIB DEBUG default - priv_set_local_sdp sofsip_cli[1558]: GLIB DEBUG default - priv_update_tx_elements sofsip_cli[1558]: GLIB MESSAGE default - No RTP destination available (r_sdp=0x85a6c). sofsip> priv_cb_pipeline_bus: Error: Could not open audio device for playback. Device is being used by another application. ~ $ sh: turning off NDELAY mode ~ $ ===========================
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