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Summary:ASTERISK-16586: Video RTP is not sended to originating SIP extension when using IAX2 to interconnect both servers
Reporter:Lorenzo Boffelli (lboff)Labels:
Date Opened:2010-08-18 06:24:56Date Closed:
Priority:MinorRegression?No
Status:Open/NewComponents:Channels/chan_iax2
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 20100818.zip
( 1) asterisk.log.tar.gz
Description:I discovered now the issue 0012902 but it has been declared closed and I can not add comments on it.
I am having the same problem described in issue 0012902 on asterisk 1.4.33

Scenario:
SIP(301)---*1---IAX(Head)---------*2---IAX(Branch)---SIP(401)

301 exec a video call passing through the IAX trunk to 401. Only 401 sees 301 video. Video RTP packets are sent by 401 but not get from Asterisk 2.
Same things happen is the caller is 401.
videosupport=yes configured on both sip and iax.
Details on SIP and IAX configuration reported below.

****** ADDITIONAL INFORMATION ******

Asterisk 1 SIP 301 configuration
--------------------------------
[301]
type=friend
secret=12345
record_out=Never
record_in=Never
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=lisa.green@system.lan
host=dynamic
dtmfmode=rfc2833
dial=SIP/301
context=from-internal
canreinvite=no
callgroup=
callerid=device <301>
accountcode=
call-limit=50
disallow=all
allow=alaw
allow=h264

Asterisk 1 IAX configuration
--------------------------------
[branch]
host=dynamic
qualify=yes
secret=branch
type=peer
username=
disallow=all
allow=alaw
allow=h264

[head]
context=from-internal
host=192.168.101.174
secret=head
type=user
disallow=all
allow=alaw
allow=h264

Asterisk 2 SIP 401 configuration
--------------------------------
[401]
type=friend
secret=12345
record_out=Never
record_in=Never
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=mike.red@system.lan
host=dynamic
dtmfmode=rfc2833
dial=SIP/401
context=from-internal
canreinvite=no
callgroup=
callerid=device <401>
accountcode=
call-limit=50
disallow=all
allow=alaw
allow=h264

Asterisk 2 IAX configuration
--------------------------------
[branch]
context=from-internal
host=192.168.101.169
secret=branch
type=user
disallow=all
allow=alaw
allow=h264

[head]
host=dynamic
qualify=yes
secret=head
type=peer
username=
disallow=all
allow=alaw
allow=h264
Comments:By: Lorenzo Boffelli (lboff) 2010-08-18 06:44:38

The issue is always reproducible.

The attached file 20100818 contains a two pcap file showing an incoming call (receives the video) and an outgoing call (does not receive the video) from SIP extension 301.

On the asteriks receiving the call (i.e. Asterisk 2) you can see that it is not processing any video RTP from the phone 401. The "Got  RTP packet" are only related to the voice packet (Type 08)

Sent RTP packet to    10.1.1.21:11794 (type 99,seq 012427,ts 266940,len 000658)
Sent RTP packet to    10.1.1.21:11792 (type 08,seq 064245,ts 616640,len 000160)
Got  RTP packet from  10.1.1.21:11792 (type 08,seq 004994,ts 623360,len 000160)
Sent RTP packet to    10.1.1.21:11792 (type 08,seq 064246,ts 616800,len 000160)
Got  RTP packet from  10.1.1.21:11792 (type 08,seq 004995,ts 623520,len 000160)
Sent RTP packet to    10.1.1.21:11794 (type 99,seq 012428,ts 266940,len 000655)
Sent RTP packet to    10.1.1.21:11794 (type 99,seq 012429,ts 266940,len 000659)
Sent RTP packet to    10.1.1.21:11794 (type 99,seq 012430,ts 266940,len 000114)
Got  RTP packet from  10.1.1.21:11792 (type 08,seq 004996,ts 623680,len 000160)
Sent RTP packet to    10.1.1.21:11792 (type 08,seq 064247,ts 616960,len 000160)
Got  RTP packet from  10.1.1.21:11792 (type 08,seq 004997,ts 623840,len 000160)
Sent RTP packet to    10.1.1.21:11792 (type 08,seq 064248,ts 617120,len 000160)
Sent RTP packet to    10.1.1.21:11794 (type 99,seq 012431,ts 266940,len 000667)
Sent RTP packet to    10.1.1.21:11794 (type 99,seq 012432,ts 266940,len 000657)
Sent RTP packet to    10.1.1.21:11794 (type 99,seq 012433,ts 266940,len 000086)
Sent RTP packet to    10.1.1.21:11792 (type 08,seq 064249,ts 617280,len 000160)
Got  RTP packet from  10.1.1.21:11792 (type 08,seq 004998,ts 624000,len 000160)
Sent RTP packet to    10.1.1.21:11792 (type 08,seq 064250,ts 617440,len 000160)
Got  RTP packet from  10.1.1.21:11792 (type 08,seq 004999,ts 624160,len 000160)
Got  RTP packet from  10.1.1.21:11792 (type 08,seq 005000,ts 624320,len 000160)
Sent RTP packet to    10.1.1.21:11792 (type 08,seq 064251,ts 617600,len 000160)
Got  RTP packet from  10.1.1.21:11792 (type 08,seq 005001,ts 624480,len 000160)
Sent RTP packet to    10.1.1.21:11792 (type 08,seq 064252,ts 617760,len 000160)
Sent RTP packet to    10.1.1.21:11794 (type 99,seq 012434,ts 266940,len 000661)
Sent RTP packet to    10.1.1.21:11794 (type 99,seq 012435,ts 266940,len 000646)


By: Leif Madsen (lmadsen) 2010-08-23 13:05:15

You will also have to provide debugging information such as console logs, SIP and IAX traces, and perhaps a pcap capture.

Also please attach your files at plain-text files to the issue as archived files are typically not opened by people looking at the issues (it's a lot more overhead).

If the files are too big, then please break them up into smaller chunks and remove information that is not directly useful.

Additionally, you're using terms like "head" which is a CVS term. Please define which versions you're using and what revision it is if not a tag.

Also, video support is limited in Asterisk, so a more simple configuration is more likely to give you better results.
By: Leif Madsen (lmadsen) 2011-01-05 15:40:32.000-0600

Suspended due to lack of feedback.
By: Lorenzo Boffelli (lboff) 2011-02-01 10:48:21.000-0600

Scenario:
SIP(301)---*1---IAX(mbox2012)------IAX(mbox2010)----*2------SIP(201)

Asterisk 1)
* SIP 301 (10.170.192.208)
* LAN interface 10.170.192.1
* WAN interface 192.168.101.254
* IAX Trunk: mbox2012

Asterisk 2)
* SIP 201 (10.170.217.201)
* LAN interface: 10.170.217.1
* WAN interface: 192.168.101.170
* IAX Trunk: mbox2010

If 201 calls 301 passing through the IAX trunk 301 see the video of 201 but 201 does not see the video of 301
If 301 calls 201 passing through the IAX trunk 201 see the video of 301 but 301 does not see the video of 201
Inother words, only the called see the video.
Both asterisk are running release 1.4.33.1
The issues is always reproducible.

asterisk.log.tar.gz contains the asterisk logs/debug while calling from 201 to 301.
asterisk-1.log if related to Asterik 1 system
asterisk-2.log if related to Asterik 2 system


By: Slava Bendersky (volga629) 2013-08-13 19:24:33.135-0500

Hello Everyone,
I can provide more information.

My setup.

SIP --- IAX trunk --- SIP

Audio travel no problem, but video get rejected right a way.




Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: NEW    
  Timestamp: 00120ms  SCall: 19548  DCall: 00000 [Ip:4569]
  VERSION         : 2
  CALLED NUMBER   : 24005
  CODEC_PREFS     : (h264|g729)
  CALLING NUMBER  : 102
  CALLING PRESNTN : 0
  CALLING TYPEOFN : 0
  CALLING TRANSIT : 0
  CALLING NAME    : Test
  LANGUAGE        : en
  USERNAME        : netlab-ott
  FORMAT          : 2097408
  FORMAT2         : Unknown
  CAPABILITY      : 2097408
  CAPABILITY2     : Unknown
  ADSICPE         : 2
  DATE TIME       : 2013-08-13  19:11:46
  CALLTOKEN       : 51 bytes

Rx-Frame Retry[ No] -- OSeqno: 009 ISeqno: 011 Type: VIDEO   Subclass: 0
  Timestamp: 12149ms  SCall: 16417  DCall: 00294 [Ip:4569]
Tx-Frame Retry[-01] -- OSeqno: 011 ISeqno: 010 Type: IAX     Subclass: ACK    
  Timestamp: 12149ms  SCall: 00294  DCall: 16417 [Ip:4569]
[2013-08-14 02:22:16] WARNING[4219][C-00000011]: res_rtp_asterisk.c:2640 ast_rtp_write: Don't know how to send format unknown packets with RTP
Tx-Frame Retry[000] -- OSeqno: 011 ISeqno: 010 Type: CONTROL Subclass: SRCUPDT