[Asterisk][Cisco] Ignoring video- 10089 process_sdp

Hello all ,

I write this post because i have a problem with my config asterisk and after many test and research = > no solution

software / config:
ver. Aserisk : 11.13
softphone : Jitsi 2.5.5227
Telepresence : Cisco Tandberg C60

test and problem:
1 - Jitsi call to Jitsi result is OK

2 - Call Tandberg C60 to Jitsi = > Cisco can send video but not receive

Error : WARNING [ 12978 ] [C - 00000007 ] chan_sip.c : Ignoring offer Because video stream port number is zero

3 - Call Jitsi to Tandberg C60 = > no video

Error : WARNING [ 12978 ] [C - 0000002d ] : chan_sip.c : 10089 process_sdp : Ignoring offer Because video stream port number is zero

Precision:

The Cisco Tandberg C60 use SIP and is registred.
Only video is reject, audio is ok!

Codec:
G711 u-law
G711 A-law
G722
H264

Sip.conf:
[general]

autocreate_context = xivo-initconfig autocreate_type = friend timer1 = 500 autocreate_prefix = apdpLFtjny buggymwi = no callerid = xivo sipdebug = no dumphistory = no recordhistory = no callevents = yes tos_sip = EF tos_audio = EF tos_video = EF t38pt_udptl = no t38pt_usertpsource = no g726nonstandard = no disallow = all allow = ulaw allow = alaw allow = g722 allow = h264 t1min = 100 relaxdtmf = yes rfc2833compensate = yes compactheaders = no rtptimeout = 0 rtpholdtimeout = 0 rtpkeepalive = 0 directrtpsetup = no srvlookup = no pedantic = no minexpiry = 60 maxexpiry = 3600 defaultexpiry = 120 registertimeout = 20 registerattempts = 0 notifyringing = yes notifyhold = yes allowtransfer = no maxcallbitrate = 1368 jbenable = no jbforce = no jbmaxsize = 200 jbresyncthreshold = 1000 jblog = no jbimpl = fixed context = default nat = no dtmfmode = rfc2833 qualify = no useclientcode = no progressinband = never mohinterpret = default vmexten = *98 trustrpid = no insecure = no rtcachefriends = yes rtupdate = no rtsavesysname = no rtautoclear = no udpbindaddr = 0.0.0.0 tcpenable = no tlsenable = no tlscadir = /var/lib/asterisk/certs/cadir tlsdontverifyserver = no tos_text = CS3 cos_sip = 7 cos_audio = 7 cos_video = 7 cos_text = 2 session-expires = 600 session-minse = 90 session-refresher = uas notifycid = yes callcounter = no bindport = 5060 allowsubscribe = yes allowoverlap = yes promiscredir = no autodomain = no allowexternaldomains = yes usereqphone = no realm = xivo alwaysauthreject = no useragent = XiVO PBX autocreatepeer = persist allowguest = yes externrefresh = 10 matchexterniplocally = no notifymimetype = application/simple-message-summary autoframing = yes prematuremedia = yes authfailureevents = no dynamic_exclude_static = no shrinkcallerid = yes regextenonqualify = no timert1 = 500 timerb = 32000 directmedia = no ignoresdpversion = yes use_q850_reason = no snom_aoc_enabled = no subscribe_network_change_event = yes domainsasrealm = no textsupport = yes videosupport = always auth_options_requests = no encryption = no transport = udp language = fr_FR sendrpid = no ignoreregexpire = no match_auth_username = no mwiexpiry = 3600 qualifyfreq = 60 qualifygap = 100 qualifypeers = 1

[jz2j6j] (Cisco C60)

amaflags = default videosupport = yes maxcallbitrate = 1368 call-limit = 10 host = dynamic language = fr_FR context = default callerid = "Cisco C40" <248> secret = ON4PHP type = friend subscribemwi = no mohsuggest = default nat = force_rport,comedia setvar = XIVO_USERID=1166 disallow = all allow = ulaw allow = alaw allow = h264 setvar = PICKUPMARK=248%default setvar = TRANSFER_CONTEXT=default cc_agent_policy = never cc_monitor_policy = never

[e6169k] b[/b]

amaflags = default videosupport = yes maxcallbitrate = 1368 call-limit = 10 host = dynamic language = fr_FR context = default callerid = "Florent MELON" <201> secret = R7RTD6 type = friend subscribemwi = no mohsuggest = default allowtransfer = 1 nat = force_rport,comedia setvar = XIVO_USERID=1164 disallow = all allow = alaw allow = ulaw allow = h264 setvar = PICKUPMARK=201%default setvar = TRANSFER_CONTEXT=default callgroup = 0 cc_agent_policy = never cc_monitor_policy = never

I hope you can help me.
if necessary I can provide more info

Thanks,
Paddman

The peer is refusing H.264. Allow a video ccdec that is prepared to use.

When the remote system uses port zero, it means it is acknowledging that you did offer video media, but is refusing to receive video from you.

Thanks,

However, it’s supposed to use H.264
http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/telepresence-codec-c60/data_sheet_c78-628595.html

I tried to change codec with H263 + , H263, and actually I do not have the error message.
But no video display! :frowning:

Yesterday, I restarted an old server with Asterisk 1.18.16
The video it’s okay (H.264) but the resolution received by Cisco is: QCIF (176X144)
We can’t change this because Cisco configures itself resolution.
H.263 / H263+ idem to Asterisk 11.13

Have you any ideas?

Hello,
I think the problema is not on the códec itself. The problem comes on how asterisk and cisco defines that códec in the rtpmap.
I do not know how to solve, but hope this is a clue for someone more.
Regards