[Asterisk-video] (continuing) Ignoring video stream offer because port number is zero v 11.3.0

[CALLEJA-ALBILLOS, Franck (Audis-Networking)]

NOTE : I'm sorry if this message is sent twice, as I sent it first before
confirming my subscription, and I do not know if it has been delivered.
Again, if you receive it twice, please acceept my apologizes.

----------------------
Hello,

I'm following up this threadŠ because I encounter the same problem as
described in the same release of Asterisk.

I have 2 users using X-Lite with h263 support trying to setup a call. I get
the same error message :

  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/101
    -- SIP/101-00000040 is ringing
[May 27 16:04:54] WARNING[23143][C-000000af]: chan_sip.c:10141 process_sdp:
Ignoring video stream offer because port number is zero
    -- SIP/101-00000040 answered SIP/100-0000003f
    -- Locally bridging SIP/100-0000003f and SIP/101-00000040
    -- Locally bridging SIP/100-0000003f and SIP/101-00000040
    -- Locally bridging SIP/100-0000003f and SIP/101-00000040
    -- Locally bridging SIP/100-0000003f and SIP/101-00000040

I can try to send Video from both X-lite (there's no error message when
activating video), but the opposite do not see it.

Khalid,
Did you solve your problem ?

When doing sip show settings :
thebox-lab2*CLI> sip show settings


Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5060
  TCP SIP Bindaddress:    Disabled
  TLS SIP Bindaddress:    Disabled
  Videosupport:           Yes
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        Off
  Match Auth Username:    No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  No
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          BOX2
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX 11.3.0
  SDP Session Name:       Asterisk PBX 11.3.0
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Trust RPID:             No
  Send RPID:              No
  Legacy userfield parse: No
  Send Diversion:         Yes
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     Off
  Call Events:            Off
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          -1
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS0
  IP ToS RTP audio:       CS0
  IP ToS RTP video:       CS0
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No

Network Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost:             <none>
  Externaddr:             (null)
  Externrefresh:          10

Global Signalling Settings:
---------------------------
  Codecs:                 (gsm|ulaw|alaw|h263|testlaw)
  Codec Order:            none
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Sub. min duration       60 secs
  Sub. max duration:      3600 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set>
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     Yes
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:   UDP
  Context:                public
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            Auto (No)
  DTMF:                   rfc2833
  Qualify:                0
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        Never
  Language:
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   asterisk



Thanks for your help.


Regards,

Franck C.




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