|Summary:||ASTERISK-13970: SIP video phone requires numberofmediastreams > 4 (chan_sip.c)|
|Date Opened:||2009-04-16 14:58:22||Date Closed:||2011-07-26 14:15:52|
|Environment:||Attachments:||( 0) asterisk_CLI_log.txt|
( 1) IPs.txt
( 2) SIP_messages_SIP_Proxy.doc
|Description:||I have a SIP video phone that I need to connect to Asterisk 1.4. The video phone sends a special invite message. The invite message includes an SDP part with 4 m blocks (media blocks).|
Since asterisk 1.4 can handle a max of 2 m blocks, this requires making a tiny change on the ./channels/chan_sip.c file. The change will modify the following line:
if (numberofmediastreams > 2)
to be changed to:
if (numberofmediastreams > 4)
I implemented the change on Asterisk 1.4. It was successful.
|Comments:||By: Tilghman Lesher (tilghman) 2009-04-16 16:48:38|
I need to know what model the phone is and what the 4 media streams are. A SIP trace would be helpful in this regard.
By: moshili (moshili) 2009-04-16 18:13:18
The video phone is called VIPR (VIrtual Presence). It is a product of Aastra company. Following are links to data sheets of couple of models:
I attached the SIP messages trace taken from the CLI of Asterisk (Asterisk examines INVITE and finds 1 audio stream + 3 video streams therefore it quits).
Additionally, I attached a SIP log showing the SIP messages on the SIP server (the SIP proxy server that the video phone uses).
Finally, I attached a clarification for what each IP corresponds to.
Please let me know if any other info is needed.
By: moshili (moshili) 2009-05-13 10:59:02
Is there any other feedback needed? Can you please implement/tweak the fix?
By: moshili (moshili) 2009-06-04 10:16:56
Can you please update me with the status? Also, can you please port the tweak to 1.6.0.x and 1.6.1.x?
By: Leif Madsen (lmadsen) 2009-09-18 08:58:11
This fix can't go in as provided, because the issue is actually larger than what it appears on your end. The fix will resolve the issue for you in your environment, but could break other environments, or not fix all situations.
I'm marking this as Acknowledged as the developers are aware of this issue, but the fix will be larger than your simple tweak.
By: Leif Madsen (lmadsen) 2011-07-26 14:15:46.659-0500
Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
Additionally I'm suspending this issue as it is significantly old, and no one seems interested in writing the more correct (and complex) method.