[Freeswitch-users] X-Lite and Video
peter at uringme.com
peter at uringme.com
Thu Jan 5 21:20:02 MSK 2012
For anyone who may have the same issue in the future, I did finally get it to work.
I had to not only add the codecs to global_codec_prefs, but add it to outbound_codec_prefs as well (then I restarted FS -- don't know if a reloadxml would have worked or not). Here's mine now:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H263,H264,H263-1998"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,H263,H264,H263-1998"/>
That got me a video call between registered extensions. However, the video was blank even though I had full audio. tcpdump showed that video data was being sent and received, but both video screens said: "Waiting for video". I tried a couple things like firewalls and NAT ICE/STUN on my own side, then finally tried removing H.263-1998 from both X-Lites to force H.263, and that worked. I had two-way video at that point. My X-Lite 4 generic doesn't have H.264 support, so I haven't tested that.
--- On Wed, 1/4/12, peter at uringme.com <peter at uringme.com> wrote:
From: peter at uringme.com <peter at uringme.com>
Subject: [Freeswitch-users] X-Lite and Video
To: freeswitch-users at lists.freeswitch.org
Date: Wednesday, January 4, 2012, 2:52 PM
(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 at 32000h,G7221 at 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 at 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 at 68.202.69.172:50350>
To: <sip:7778 at test>
From: "Peter Test"<sip:7777 at 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 at 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 at 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 at 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 at 204.13.175.89>;tag=ZeZUav5XXat5e
To: <sip:7778 at 68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a>
Call-ID: fe64b2f5-b1a6-122f-a187-00144f49eecc
CSeq: 22515274 INVITE
Contact: <sip:mod_sofia at 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 at 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".
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