<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;">(Apologies to the list owner for originally sending this to the wrong address)<br><br><br>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:<br><br> <extension name="7777"><br> <condition field="destination_number" expression="^7777$"><br> <action application="bridge" data="sofia/external/7777%test"/><br> </condition><br> </extension><br>(and vice-versa for 7778).<br><br>I can dial between them just fine for audio calls -- bidirectional audio, etc, no problem.<br><br>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:<br> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263,H264,H263-1998"/><br><br>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.<br><br>INVITE from freeswitch console:<br> ------------------------------------------------------------------------<br> INVITE sip:7778@test SIP/2.0<br> Via: SIP/2.0/UDP 192.168.1.7:50350;branch=z9hG4bK-d8754z-9ea3618224c5cf23-1---d8754z-;rport<br> Max-Forwards: 70<br> Contact: <sip:7777@68.202.69.172:50350><br> To: <sip:7778@test><br> From: "Peter Test"<sip:7777@test>;tag=a7031d83<br> Call-ID: ZDUzZGE1YjUyOTQ2ZGNmZTY0Yjc5ODA5NTE4NDAzMGQ.<br> CSeq: 1 INVITE<br> Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br> Content-Type: application/sdp<br> Supported: replaces<br> User-Agent: X-Lite 4 release 4.1 stamp 63214<br> Content-Length: 681<br><br> v=0<br> o=- 12970176286658638 1 IN IP4 192.168.1.7<br> s=CounterPath X-Lite 4.1<br> c=IN IP4 192.168.1.7<br> t=0 0<br> a=ice-ufrag:20fef6<br> a=ice-pwd:0a03863684bc5f16a9c862dcdccdd8eb<br> m=audio 58632 RTP/AVP 107 0 8 101<br> a=rtpmap:107 BV32/16000<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-15<br> a=sendrecv<br> a=candidate:1 1 UDP 659136 192.168.1.7 58632 typ host<br> a=candidate:1 2 UDP 659134 192.168.1.7 58633 typ host<br> m=video 50994 RTP/AVP 34 115<br> a=rtpmap:34
H263/90000<br> a=fmtp:34 QCIF=2;CIF=2;VGA=2<br> a=rtpmap:115 H263-1998/90000<br> a=fmtp:115 QCIF=2;CIF=2;VGA=2;I=1;J=1;T=1<br> a=sendrecv<br> a=candidate:1 1 UDP 659136 192.168.1.7 50994 typ host<br> a=candidate:1 2 UDP 659134 192.168.1.7 50995 typ host<br><br>I do see freeswitch seeing the audio and video codecs:<br>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]<br>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<br>2012-01-04 13:44:37.188657 [DEBUG] switch_core_state_machine.c:343 (sofia/external/7777@test) State NEW<br>2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4797 Set 2833 dtmf send/recv payload to 101<br>2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4856 Video Codec Compare [H263:34]/[H263:34]<br><br>However, when
freeswitch starts the bridge and calls the far-end party, it doesn't send along video information in the INVITE:<br><br> INVITE sip:7778@68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a SIP/2.0<br> Via: SIP/2.0/UDP 204.13.175.89:5080;rport;branch=z9hG4bK8jm56tcmZ6p6j2012-01-04 <br> Max-Forwards: 69<br> From: "Peter Test" <sip:7777@204.13.175.89>;tag=ZeZUav5XXat5e<br> To: <sip:7778@68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a><br> Call-ID: fe64b2f5-b1a6-122f-a187-00144f49eecc<br> CSeq: 22515274 INVITE<br> Contact: <sip:mod_sofia@204.13.175.89:5080><br> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br> Supported: timer, precondition, path,
replaces<br> Allow-Events: talk, hold, refer<br> Content-Type: application/sdp<br> Content-Disposition: session<br> Content-Length: 207<br> X-FS-Support: update_display<br> Remote-Party-ID: "Peter Test" <sip:7777@204.13.175.89>;party=calling;screen=yes;privacy=off<br><br> v=0<br> o=FreeSWITCH <span class="yshortcuts" id="lw_1325706674_0">1325690959</span> 1325690960 IN IP4 204.13.175.89<br> s=FreeSWITCH<br> c=IN IP4 204.13.175.89<br> t=0 0<br> m=audio 11718 RTP/AVP 0 8 3 101 13<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-16<br> a=ptime:20<br><br><br>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.<br><br>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".</td></tr></table>