(Apologies to the list owner for originally sending this to the wrong address) I have two laptops running X-Lite 4. I have them registered to a FreeSwitch server (latest git) as extensions 7777 and 7778. I have a dialplan for each (quick and dirty) that just bridges them when one is dialed from the other: <extension name="7777"> <condition field="destination_number" expression="^7777$"> <action application="bridge" data="sofia/external/7777%test"/> </condition> </extension> (and vice-versa for 7778). I can dial between them just fine for audio calls -- bidirectional audio, etc, no problem. I'm trying to get video going. Both X-Lites have H.263 and H.263-1998 enabled in their settings. Freeswitch has the following in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263,H264,H263-1998"/> When I try to make a video call from one extension to the other, the calling extension seems to think it's in video, but the called extension doesn't. INVITE from freeswitch console: ------------------------------------------------------------------------ INVITE sip:7778@test SIP/2.0 Via: SIP/2.0/UDP 192.168.1.7:50350;branch=z9hG4bK-d8754z-9ea3618224c5cf23-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:7777@68.202.69.172:50350> To: <sip:7778@test> From: "Peter Test"<sip:7777@test>;tag=a7031d83 Call-ID: ZDUzZGE1YjUyOTQ2ZGNmZTY0Yjc5ODA5NTE4NDAzMGQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 681 v=0 o=- 12970176286658638 1 IN IP4 192.168.1.7 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.1.7 t=0 0 a=ice-ufrag:20fef6 a=ice-pwd:0a03863684bc5f16a9c862dcdccdd8eb m=audio 58632 RTP/AVP 107 0 8 101 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.7 58632 typ host a=candidate:1 2 UDP 659134 192.168.1.7 58633 typ host m=video 50994 RTP/AVP 34 115 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=2;VGA=2 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=2;CIF=2;VGA=2;I=1;J=1;T=1 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.7 50994 typ host a=candidate:1 2 UDP 659134 192.168.1.7 50995 typ host I do see freeswitch seeing the audio and video codecs: 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4683 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:2800 Set Codec sofia/external/7777@test PCMU/8000 20 ms 160 samples 64000 bits 2012-01-04 13:44:37.188657 [DEBUG] switch_core_state_machine.c:343 (sofia/external/7777@test) State NEW 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4797 Set 2833 dtmf send/recv payload to 101 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4856 Video Codec Compare [H263:34]/[H263:34] However, when freeswitch starts the bridge and calls the far-end party, it doesn't send along video information in the INVITE: INVITE sip:7778@68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a SIP/2.0 Via: SIP/2.0/UDP 204.13.175.89:5080;rport;branch=z9hG4bK8jm56tcmZ6p6j2012-01-04 Max-Forwards: 69 From: "Peter Test" <sip:7777@204.13.175.89>;tag=ZeZUav5XXat5e To: <sip:7778@68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a> Call-ID: fe64b2f5-b1a6-122f-a187-00144f49eecc CSeq: 22515274 INVITE Contact: <sip:mod_sofia@204.13.175.89:5080> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 207 X-FS-Support: update_display Remote-Party-ID: "Peter Test" <sip:7777@204.13.175.89>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1325690959 1325690960 IN IP4 204.13.175.89 s=FreeSWITCH c=IN IP4 204.13.175.89 t=0 0 m=audio 11718 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 So, am I missing something? Do I need to use something other than "bridge" in the dialplan, or do I need to add some variables to be able to pass on the video? All I'm trying to do is make a video call between two X-Lites that are locally SIP registered to freeswitch. Because I want to record the video at some point in the future, I don't want to divert the media -- I want it streaming/passing through freeswitch. When the call is connected, the caller shows a "Waiting for video", but the called doesn't show this. When I try to start the video, it says "Failed to Start Video".