[Freeswitch-users] FreeSwithc not handling no SDP INVITE properly for video call

[Anthony Minessale]

ok then file it on jira
http://jira.freeswitch.org
It's closed for maintenance so report it tomorrow.


On Mon, Feb 28, 2011 at 4:57 PM, George Lee <whglee at gmail.com> wrote:
> I have already had this setup in my dialplan in this failure case.
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="set" data="inherit_codec=true"/>
>         <action application="set" data="proxy_media=false"/>
>
> - George
>
> On Mon, Feb 28, 2011 at 5:43 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> try
>>
>> <action application="set" data="inherit_codec=true"/>
>>
>> in your dialplan before you call the phone.
>>
>>
>>
>> On Mon, Feb 28, 2011 at 12:04 PM, George Lee <whglee at gmail.com> wrote:
>>> I pulled from the latest source tree and rebuilt FreeSwitch. The
>>> problem still persists.
>>> http://pastebin.freeswitch.org/15486
>>> Any other suggestions you could provide?
>>>
>>> Thanks,
>>> George
>>>
>>> On Fri, Feb 25, 2011 at 5:52 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> make sure you have tested on latest GIT (minutes ago) there was just a
>>>> fix to some video issues.
>>>>
>>>>
>>>> On Fri, Feb 25, 2011 at 3:56 PM, George Lee <whglee at gmail.com> wrote:
>>>>> Hi all,
>>>>>
>>>>> I am having trouble with FreeSwitch handling late codec negotiation
>>>>> for video calls.
>>>>>
>>>>> The call logs are here:
>>>>> http://pastebin.freeswitch.org/15481
>>>>>
>>>>> I have this line:
>>>>> <param name="enable-3pcc" value="proxy"/>
>>>>> added to the external sip_profile for handling late codec negotiation.
>>>>> I also have
>>>>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,H263"/>
>>>>>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,H263"/>
>>>>> in vars.xml for codec support list.
>>>>>
>>>>> The initial INVITE contains no SDP for the caller and once the callee
>>>>> answers the call, it sends the 200 OK with audio (pcma, pcmu) and
>>>>> video (H263) codec capabilities. FreeSwitch forwards the 200 OK to the
>>>>> caller without video codec capabilities. As a result, the video call
>>>>> does not establish properly.
>>>>>
>>>>> Could someone give me a pointer?
>>>>>
>>>>> Thanks,
>>>>> George
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
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>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
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>>>>
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>>>>
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>>>
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>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900