FreeSwithc not handling no SDP INVITE properly for video call
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
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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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@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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
participants (2)
-
Anthony Minessale
-
George Lee