a=fmtp missing on FS answer
Hello, I´m having some problems with códec negotiation and one way audio . I take some Sip captures and I notice that Freeswitch is not inserting * fmtp* parameters in SDP in his 200OK response. I notice that this is a common behavior. I really need that FS generates the complete SDP (with fmtp) because some devices only accept for example G729 annexb=no. I tried with the function sip_append_audio_sdp but it only accepts one a=fmtp line. “ <tr valign=top> <th align="left"> @RFC3264: SDP Offer/Answer Negotiation </th> <td> Generating and processing offers or answers. </td> <td> - "a=fmtp" parameters are not taken into account when generating or processing answer </td> </tr> ” Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso@redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net
you probably want to set verbose_sdp=true On Thu, Nov 17, 2011 at 1:17 PM, Brian West <brian@freeswitch.org> wrote:
what exactly are you doing and what are the sip traces? What firmware rev?
/b
On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote:
sip_append_audio_sdp
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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´m trying to make a call trough FS with G729 annexb=no in one side and G729 annexb=no ( the same) in the other. I have a sangoma transcoding card installed and configured, all is working fine. I use this card because in some cases I need transcoding . So I have late-negotiation=true and disable-transcoding=false. The problem is that FS answer without a=fmtp as show in the picture. Freeswitch version is 1.0 [image: sdp-problem.jpg] *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* jueves, 17 de noviembre de 2011 16:18 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer what exactly are you doing and what are the sip traces? What firmware rev? /b On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: sip_append_audio_sdp
Anthony said you probably want this: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp -Avi On Thu, Nov 17, 2011 at 10:24 PM, Camila Troncoso <ctroncoso@redvoiss.net>wrote:
I´m trying to make a call trough FS with G729 annexb=no in one side and G729 annexb=no ( the same) in the other. I have a sangoma transcoding card installed and configured, all is working fine. I use this card because in some cases I need transcoding . So I have late-negotiation=true and disable-transcoding=false. The problem is that FS answer without a=fmtp as show in the picture. Freeswitch version is 1.0
[image: sdp-problem.jpg]
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* jueves, 17 de noviembre de 2011 16:18 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer
what exactly are you doing and what are the sip traces? What firmware rev?
/b
On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote:
sip_append_audio_sdp
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
This isnt the solution because it only appends rtpmap not fmtp, I tried it. *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* jueves, 17 de noviembre de 2011 18:01 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer Anthony said you probably want this: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp -Avi On Thu, Nov 17, 2011 at 10:24 PM, Camila Troncoso <ctroncoso@redvoiss.net> wrote: I´m trying to make a call trough FS with G729 annexb=no in one side and G729 annexb=no ( the same) in the other. I have a sangoma transcoding card installed and configured, all is working fine. I use this card because in some cases I need transcoding . So I have late-negotiation=true and disable-transcoding=false. The problem is that FS answer without a=fmtp as show in the picture. Freeswitch version is 1.0 *Error! Filename not specified.**Error! Filename not specified.* *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* jueves, 17 de noviembre de 2011 16:18 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer what exactly are you doing and what are the sip traces? What firmware rev? /b On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: sip_append_audio_sdp _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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
{sip_append_audio_sdp=a=fmtp:18 annexb=no}sofia/internal/foo@bar.com Or some variation of that? /b On Nov 17, 2011, at 3:05 PM, Camila Troncoso wrote:
This isnt the solution because it only appends rtpmap not fmtp, I tried it.
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* jueves, 17 de noviembre de 2011 18:01 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer
Anthony said you probably want this:
http://wiki.freeswitch.org/wiki/Variable_verbose_sdp
-Avi
On Thu, Nov 17, 2011 at 10:24 PM, Camila Troncoso <ctroncoso@redvoiss.net> wrote:
I´m trying to make a call trough FS with G729 annexb=no in one side and G729 annexb=no ( the same) in the other. I have a sangoma transcoding card installed and configured, all is working fine. I use this card because in some cases I need transcoding . So I have late-negotiation=true and disable-transcoding=false. The problem is that FS answer without a=fmtp as show in the picture. Freeswitch version is 1.0
*Error! Filename not specified.**Error! Filename not specified.*
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* jueves, 17 de noviembre de 2011 16:18 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer
what exactly are you doing and what are the sip traces? What firmware rev?
/b
On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote:
sip_append_audio_sdp
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Yes , i tried that solution, but I force it to be annexb=no, what if the calling party has annexb=yes ( and the called party also)]? I can´t know “a priori” what codec will be used. I’m looking for a more drastic solution , where the answer form FS takes also into account the fmtp from the negotiated codec . *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* jueves, 17 de noviembre de 2011 18:34 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer {sip_append_audio_sdp=a=fmtp:18 annexb=no}sofia/internal/foo@bar.com Or some variation of that? /b On Nov 17, 2011, at 3:05 PM, Camila Troncoso wrote: This isnt the solution because it only appends rtpmap not fmtp, I tried it. *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Avi Marcus *Sent:* jueves, 17 de noviembre de 2011 18:01 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer Anthony said you probably want this: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp -Avi On Thu, Nov 17, 2011 at 10:24 PM, Camila Troncoso <ctroncoso@redvoiss.net> wrote: I´m trying to make a call trough FS with G729 annexb=no in one side and G729 annexb=no ( the same) in the other. I have a sangoma transcoding card installed and configured, all is working fine. I use this card because in some cases I need transcoding . So I have late-negotiation=true and disable-transcoding=false. The problem is that FS answer without a=fmtp as show in the picture. Freeswitch version is 1.0 *Error! Filename not specified.**Error! Filename not specified.* *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* jueves, 17 de noviembre de 2011 16:18 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] a=fmtp missing on FS answer what exactly are you doing and what are the sip traces? What firmware rev? /b On Nov 17, 2011, at 12:52 PM, Camila Troncoso wrote: sip_append_audio_sdp _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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
If its not doing what you want.. and you tried everything you know email consulting@freeswitch.org for a quote to get it to do what you want. /b On Nov 18, 2011, at 7:19 AM, Camila Troncoso wrote:
Yes , i tried that solution, but I force it to be annexb=no, what if the calling party has annexb=yes ( and the called party also)]? I can´t know “a priori” what codec will be used. I’m looking for a more drastic solution , where the answer form FS takes also into account the fmtp from the negotiated codec .
participants (4)
-
Anthony Minessale
-
Avi Marcus
-
Brian West
-
Camila Troncoso