I cannot get video support on Asterisk 1.6.1.1 or 1.6.1.6…I must be doing something stupid here. My 1.4 setup worked fine.
Clients are both Counterpath SIP phones on the same network. I get the “failed to start video” error on both ends.
Here’s my sip.conf:
[general]
…
videosupport=always
maxcallbitrate=384
…
basic-options
dtmfmode=rfc2833
context=domain.org
type=friend
natted-phone
nat=yes
canreinvite=no
host=dynamic
videosupport=yes
qualify=yes
maxcallbitrate=384
public-phone
nat=no
canreinvite=yes
fring-phone
nat=yes
host=dynamic
my-codecs
disallow=all
allow=h263p
allow=ulaw
allow=ilbc
allow=gsm
allow=g723
allow=g729
allow=alaw
203
callerid=“jstrope” <9612542525>
332
callerid=“Test Phone” <332>
Here’s a sip debug from the start to of the video invite (192.168.0.5 is Asterisk):
<— SIP read from UDP://192.168.0.20:11626 —>
INVITE sip:332@192.168.0.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:11626;branch=z9hG4bK-d8754z-25cd8c3127bc3e94-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:203@192.168.0.20:11626;rinstance=ed922796f5aa52a5
To: "Test Phone"sip:332@192.168.0.5;tag=as7726ccdf
From: sip:203@192.168.0.20:11626;rinstance=ed922796f5aa52a5;tag=d4880029
Call-ID: 442d9ad602242aaa2e17e4644f0fc91d@192.168.0.5
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria Professional release 2.5.4 stamp 53962
Content-Length: 688
v=0
o=- 1 4 IN IP4 192.168.0.20
s=CounterPath Bria Professional
c=IN IP4 192.168.0.20
t=0 0
m=audio 48786 RTP/AVP 107 0 8 101
a=sendrecv
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : UdzfA3q/ TyPXxxLr 192.168.0.20 48786
m=video 12278 RTP/AVP 115 34 123 124
a=sendrecv
a=rtpmap:115 H263-1998/90000
a=fmtp:115 QCIF=2;CIF=3;I=1;J=1;T=1
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=2;CIF=3
a=rtpmap:123 H264/90000
a=fmtp:123 profile-level-id=42801e; packetization-mode=0; max-mbps=48600
a=rtpmap:124 H264/90000
a=fmtp:124 profile-level-id=42801e; packetization-mode=1; max-mbps=48600
a=alt:1 1 : JE8QNJ8D U9V6sSJ8 192.168.0.20 12278
<------------->
— (13 headers 22 lines) —
Sending to 192.168.0.20 : 11626 (NAT)
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found RTP video format 115
Found RTP video format 34
Found RTP video format 123
Found RTP video format 124
Peer audio RTP is at port 192.168.0.20:48786
Found unknown media description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Found video description format H263-1998 for ID 115
Found video description format H263 for ID 34
Found video description format H264 for ID 123
Found video description format H264 for ID 124
Capabilities: us - 0x10050f (g723|gsm|ulaw|alaw|g729|ilbc|h263p), peer - audio=0xc (ulaw|alaw)/video=0x380000 (h263|h263p|h264)/text=0x0 (nothing), combined - 0x10000c (ulaw|alaw|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.20:48786
<— Transmitting (NAT) to 192.168.0.20:11626 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.20:11626;branch=z9hG4bK-d8754z-25cd8c3127bc3e94-1—d8754z-;received=192.168.0.20;rport=11626
From: sip:203@192.168.0.20:11626;rinstance=ed922796f5aa52a5;tag=d4880029
To: "Test Phone"sip:332@192.168.0.5;tag=as7726ccdf
Call-ID: 442d9ad602242aaa2e17e4644f0fc91d@192.168.0.5
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:332@192.168.0.5
Content-Length: 0
<------------>
Audio is at 192.168.0.5 port 15120
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 192.168.0.20:11626 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.20:11626;branch=z9hG4bK-d8754z-25cd8c3127bc3e94-1—d8754z-;received=192.168.0.20;rport=11626
From: sip:203@192.168.0.20:11626;rinstance=ed922796f5aa52a5;tag=d4880029
To: "Test Phone"sip:332@192.168.0.5;tag=as7726ccdf
Call-ID: 442d9ad602242aaa2e17e4644f0fc91d@192.168.0.5
CSeq: 3 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:332@192.168.0.5
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 517629292 517629294 IN IP4 192.168.0.5
s=Asterisk PBX 1.6.1.6
c=IN IP4 192.168.0.5
t=0 0
m=audio 15120 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 115 34 123 124
<------------>
social*CLI>
<— SIP read from UDP://192.168.0.20:11626 —>
ACK sip:332@192.168.0.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:11626;branch=z9hG4bK-d8754z-4d0d5c5a8b56c5ff-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:203@192.168.0.20:11626;rinstance=ed922796f5aa52a5
To: "Test Phone"sip:332@192.168.0.5;tag=as7726ccdf
From: sip:203@192.168.0.20:11626;rinstance=ed922796f5aa52a5;tag=d4880029
Call-ID: 442d9ad602242aaa2e17e4644f0fc91d@192.168.0.5
CSeq: 3 ACK
User-Agent: Bria Professional release 2.5.4 stamp 53962
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Any ideas?