Hello all,
How do you get video calling working on Asterisk?
I have my 8945s voice calling each other and calling out etc but I get no option on the phone to make a video call.
My Asterisk version is 1.8.17.0
I have Video set to Enabled in ASterisk SIP Settings and h264,h263p,h263 and h261 ticked.
In my SEPmac.cnf.xml file I have
1
1
Show us the SDP transaction.
I have just noticed that in the logs I keep getting
Ignoring video media offer because port number is zero
Is this what you mean?
core show translation
Translation times between formats (in microseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g719 speex16 testlaw
g723 - - - - - - - - - - - - - - - - - - -
gsm - - 2 2 2001 1001 1 1001 - - 10000 3001 2 - - 1002 - - 1000
ulaw - 1001 - 1 2001 1001 1 1001 - - 10000 3001 2 - - 1002 - - 1000
alaw - 1001 1 - 2001 1001 1 1001 - - 10000 3001 2 - - 1002 - - 1000
g726aal2 - 2000 1001 1001 - 2000 1000 2000 - - 10999 4000 1001 - - 2001 - - 1999
adpcm - 1001 2 2 2001 - 1 1001 - - 10000 3001 2 - - 1002 - - 1000
slin - 1000 1 1 2000 1000 - 1000 - - 9999 3000 1 - - 1001 - - 999
lpc10 - 2999 2000 2000 3999 2999 1999 - - - 11998 4999 2000 - - 3000 - - 2998
g729 - - - - - - - - - - - - - - - - - - -
speex - - - - - - - - - - - - - - - - - - -
ilbc - 1999 1000 1000 2999 1999 999 1999 - - - 3999 1000 - - 2000 - - 1998
g726 - 2000 1001 1001 3000 2000 1000 2000 - - 10999 - 1001 - - 2001 - - 1999
g722 - 1001 2 2 2001 1001 1 1001 - - 10000 3001 - - - 1000 - - 1000
siren7 - - - - - - - - - - - - - - - - - - -
siren14 - - - - - - - - - - - - - - - - - - -
slin16 - 2001 1002 1002 3001 2001 1001 2001 - - 11000 4001 1000 - - - - - 2000
g719 - - - - - - - - - - - - - - - - - - -
speex16 - - - - - - - - - - - - - - - - - - -
testlaw - 1001 2 2 2001 1001 1 1001 - - 10000 3001 2 - - 1002 - - -
No I said SDP transaction. You need to enable the sip debug to get that.
Is this what you mean?
I did a “sip set debug ip 192.168.249.156”
<— SIP read from UDP:192.168.249.156:5060 —>
INVITE sip:582@192.168.249.18;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.249.156:5060;branch=z9hG4bK0354e27a
From: “Mark Pywell” sip:572@192.168.249.18;tag=8478acece1bd00825a8a579c-7822f5e1
To: sip:582@192.168.249.18
Call-ID: 8478acec-e1bd0007-340ac6bf-37f9e1aa@192.168.249.156
Max-Forwards: 70
Date: Thu, 22 Aug 2013 15:01:40 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP8945/9.4.1
Contact: sip:572@192.168.249.156:5060;transport=udp;video
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 691
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 12552 0 IN IP4 192.168.249.156
s=SIP Call
t=0 0
m=audio 10012 RTP/AVP 0 8 18 102 9 116 101
c=IN IP4 192.168.249.156
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 10014 RTP/AVP 97
c=IN IP4 192.168.249.156
b=TIAS:2000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200
a=imageattr:97 send [x=640,y=480] [x=640,y=360] [x=352,y=288] [x=176,y=144] recv [x=640,y=480]
a=sendrecv
<------------->
<— SIP read from UDP:192.168.249.156:5060 —>
INVITE sip:582@192.168.249.18;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.249.156:5060;branch=z9hG4bK296a987f
From: “Mark Pywell” sip:572@192.168.249.18;tag=8478acece1bd00825a8a579c-7822f5e1
To: sip:582@192.168.249.18
Call-ID: 8478acec-e1bd0007-340ac6bf-37f9e1aa@192.168.249.156
Max-Forwards: 70
Date: Thu, 22 Aug 2013 15:01:40 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP8945/9.4.1
Contact: sip:572@192.168.249.156:5060;transport=udp;video
Authorization: Digest username=“572”,realm=“asterisk”,uri="sip:582@192.168.249.18;user=phone",response=“31049d7c4feb45a43eeda15e6ebb07b5”,nonce=“5ff61107”,algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 691
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 12552 0 IN IP4 192.168.249.156
s=SIP Call
t=0 0
m=audio 10012 RTP/AVP 0 8 18 102 9 116 101
c=IN IP4 192.168.249.156
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 10014 RTP/AVP 97
c=IN IP4 192.168.249.156
b=TIAS:2000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200
a=imageattr:97 send [x=640,y=480] [x=640,y=360] [x=352,y=288] [x=176,y=144] recv [x=640,y=480]
a=sendrecv
<------------->
e[Ksentinel*CLI>
e[0K— (19 headers 24 lines) —
e[Ksentinel*CLI>
e[0KSending to 192.168.249.156:5060 (no NAT)
e[Ksentinel*CLI>
e[0KUsing INVITE request as basis request - 8478acec-e1bd0007-340ac6bf-37f9e1aa@192.168.249.156
e[Ksentinel*CLI>
e[0KFound peer ‘572’ for ‘572’ from 192.168.249.156:5060
e[Ksentinel*CLI>
e[0K == Using SIP RTP TOS bits 184
e[Ksentinel*CLI>
e[0K == Using SIP RTP CoS mark 5
e[Ksentinel*CLI>
e[0KFound RTP audio format 0
e[Ksentinel*CLI>
e[0KFound RTP audio format 8
e[Ksentinel*CLI>
e[0KFound RTP audio format 18
e[Ksentinel*CLI>
e[0KFound RTP audio format 102
e[Ksentinel*CLI>
e[0KFound RTP audio format 9
e[Ksentinel*CLI>
e[0KFound RTP audio format 116
e[Ksentinel*CLI>
e[0KFound RTP audio format 101
e[Ksentinel*CLI>
e[0KFound audio description format PCMU for ID 0
e[Ksentinel*CLI>
e[0KFound audio description format PCMA for ID 8
e[Ksentinel*CLI>
e[0KFound audio description format G729 for ID 18
e[Ksentinel*CLI>
e[0KFound audio description format L16 for ID 102
e[Ksentinel*CLI>
e[0KFound audio description format G722 for ID 9
e[Ksentinel*CLI>
e[0KFound audio description format iLBC for ID 116
e[Ksentinel*CLI>
e[0KFound audio description format telephone-event for ID 101
e[Ksentinel*CLI>
e[0KFound RTP video format 97
e[Ksentinel*CLI>
e[0KFound video description format H264 for ID 97
e[Ksentinel*CLI>
e[0KCapabilities: us - 0x20000e (gsm|ulaw|alaw|h264), peer - audio=0x950c (ulaw|alaw|g729|ilbc|g722|slin16)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x20000c (ulaw|alaw|h264)
e[Ksentinel*CLI>
e[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
e[Ksentinel*CLI>
e[0KPeer audio RTP is at port 192.168.249.156:10012
e[Ksentinel*CLI>
e[0KPeer video RTP is at port 192.168.249.156:10014
e[Ksentinel*CLI>
e[0KLooking for 582 in from-internal (domain 192.168.249.18)
Yes, but the complete debug, that only show one phone and seems the video port for it is in 10014