Dear Community!
I'm having a problem with not getting video calls working with CUCM version 11.5. Audio works.
System setup is:
MicroSIP -> 3rd party SIP server -> CUCM -> MicroSIP
Initial setup was no SDP sent from CUCM to callee - only in OK (INVITE). CUCM only sent "audio" and "video" in contact parameter, else no trace of video capabilities.
I suspect MicroSIP then not "daring" to set video port number and only audio get a port number different from 0 in the OK (INVITE) from MicroSIP.
Observing Wireshark, 3rd party SIP server sends 3 audio and 4 video codecs in SDP to CUCM. Both audio and video have port numbers different from 0.
But then, this is not passed on to callee MicroSIP.
CUCM has port 0 for video in ACK, so no video. This must be due to MicroSIP prior to this has answered with OK (INVITE) where SDP has port 0 for video.
Same issue when both MicroSIP clients are registred on CUCM (internal in CUCM):
MicroSIP -> CUCM -> MicroSIP
I then configured CUCM to send SDP in initial INVITE, to make sure callee MicroSIP gets the video capabilities.
The only way to enable SDP in initial INVITE was to set Media Termination Point Required for the phone.
But CUCM then only send one audio codec to callee.
How can I make sure CUCM passes on video capabilities to the callee MicroSIP?
Settings (some listed which I think is relevant):
System/Device Pool: "default"
Location: "Hub_none"
System/Region Information/Region: Regin "default"
Maximum Session Bit Rate for Video Calls: 2000 kbps
System/Location Info/Location: "Hub_none"
Links - Bandwidth Between Hub_None and Adjacent Locations: Not possible to add, "already configured"
Intra-location - Bandwidth for Devices Within This Location: Video Bandwidth Unlimited.
Location RSVP Settings: No Reservation.
Device/Trunk: "3rd party"
Device Pool: "Default"
Location: "Hub_none"
Media Termination Point Required: True
Retry Video Call as Audio: True
SIP Profile: "MySIP"
Device/Device Settings/SIP Profile: "MySIP"
SDP Information:
SDP Session-level Bandwidth Modifier for Early Offer and Re-invites: TIAS and AS
Trunk Specific Configuration:
Early Offer support for voice and video calls: Mandatory (insert MTP if needed)
Device/Phone:
Device Information:
Common Phone Profile: Standard common phone profile
Product Specific Configuration Layout:
Video Calling: Enabled
Location: "Hub_none"
Protocol Specific Information:
MTP Preferred Originating Codec: 711ulaw (same only codec sent in SDP to callee. Can change to 711alaw, which is then the only one in SDP)
Media Termination Point Required: True (else no SDP in initial INVITE from CUCM).
SIP Profile: "MySIP"
