sushil verma wrote: I don't know if this could help, but it worked for me :-)
> Tried an incoming call using sofsip_cli. No rtp going out. Call is established and active. Following are the logs. Any solution? [...] > ** (<unknown>:4559): DEBUG: farsight_netsocket_stun_map > ** (<unknown>:4559): DEBUG: server:(null) domain:134.177.144.34. > UA: answering to the offer received from "Sushil Kumar" <sip:3...@scs500.us.nortel.com>;tag=1fcd30d7 > sofsip> ** (<unknown>:4559): DEBUG: cb_stun_state: stun_error (8) > ** (<unknown>:4559): DEBUG: cb_stun_state: error in communication with the STUN server This could be a first problem, you didn't define a STUN server, however it tries to use it. In file ssc_sip.c, function priv_create_ssc_media, around line 300: if (res_impl) { g_object_set(G_OBJECT(res_impl), "stun-server", conf->ssc_stun_server, NULL); g_object_set(G_OBJECT(res_impl), "stun-domain", userdomain, NULL); } If you don't define conf->ssc_stun_server then it's NULL, but userdomain is never NULL, so in ssc_media_gst.c, function priv_setup_rtpelements, near line 470: if (self->sm_stun_server || self->sm_stun_domain) { self->sm_netsocket = NULL; netsocket_type = FARSIGHT_NETSOCKET_STUN_TYPE; netsocket = g_object_new (netsocket_type, "sockfd", pri_sockfd, "aux-sockfd", aux_sockfd, "server", self->sm_stun_server, "domain", self->sm_stun_domain, NULL); [...] It always uses STUN. I fixed it in ssc_sip.c changing the code to: if (res_impl) { g_object_set(G_OBJECT(res_impl), "stun-server", conf->ssc_stun_server, NULL); if(conf->ssc_stun_server){ g_object_set(G_OBJECT(res_impl), "stun-domain", userdomain, NULL); } else { g_object_set(G_OBJECT(res_impl), "stun-domain", NULL, NULL); } } You can fix it in ssc_media_gst.c if you prefer. > ** (<unknown>:4559): DEBUG: priv_cb_ready > ** Message: Created audio sink of type 'ALSA' for playback. > > (<unknown>:4559): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `AlsaSPDIFSink' has no property named `latency-time' > > (<unknown>:4559): GLib-GObject-WARNING **: IA__g_object_set_valist: object class `AlsaSPDIFSink' has no property named `buffer-time' This one could be a second problem. It looks for the gst element alsasink but the first one that it finds is AlsaSPDIFSink but it's the wrong one. In the file ssc_media_gst_utils.c the function ssc_media_create_audio_sink near line 205 checks only if the string alsa is in the longname, so for it AlsaSPDIFSink and AlsaSink are the same: if(strstr(klass_tags, "Sink") && strstr(klass_tags, "Audio") && strstr(gst_element_factory_get_longname(factory), type)){ res = gst_element_factory_create(factory, "audiosink"); break; } I fixed it in a quick and dirty way like this (I'll have to think for a better way though :-)): if (strstr(klass_tags, "Sink") && strstr(klass_tags, "Audio")){ if(strstr(gst_element_factory_get_longname(factory), type) && !strstr(gst_element_factory_get_longname(factory), "S/PDIF ALSA") ){ res = gst_element_factory_create(factory, "audiosink"); break; } } > priv_cb_pipeline_bus: Error: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer. Never had this error, so I don't know how to deal with it. Hope this helps. Regards, Agostino ------------------------------------------------------------------------------ Sell apps to millions through the Intel(R) Atom(Tm) Developer Program Be part of this innovative community and reach millions of netbook users worldwide. Take advantage of special opportunities to increase revenue and speed time-to-market. Join now, and jumpstart your future. http://p.sf.net/sfu/intel-atom-d2d _______________________________________________ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel