Hi all, I want to do a special SIP using and I dont know how to configure it with: originate/bridge/transfer no_media and so on May be it is not possible with freeswitch? Here comes scenario: - one sender device that sends multicast RTP stream to network - one or more receiver devices that only receive RTP data from network The codec is a proprietary one (41), freeswitch can ignore all RTP data. Sender has to get attributs: u=file:/C:/dap/123.wav c=IN IP4 239.194.0.14 t=0 0 m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=recvonly b=AS:128 Receiver has to get attributs: c=IN IP4 239.193.201.69 t=0 0 a=specialExtension:2 a=type:broadcast m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=sendonly Is it possible to use freeswitch API to do this? Many thanks in advance, Thomas Sluschny
Hello, unfortunately no one answered til now. So i ask a bit different: 1. Is it possible to create a call to a SIP device where i can define the MessageBody or SDP field on my own (as mentioned in first mail)? I want to INVITE one sender (plays an announcement) and many receivers. These INVITEs should be independent, so if one receiver is out of order all others can be announced anyway. 2. Is it possible to invite a SIP device independend from others? I tested: pacall device1; pacall device2; uuid_bridge sip_uuid1 sip_uuid2 -> this is a little bit sick, i know ;) originate sofia/domain/device1 device2 -> but this works only with 2 devices AND the INVITEs go out successively Has this "media" command anything to do with it? Hope this clears more my problem. Thanks, Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Sluschny, Thomas Gesendet: Donnerstag, 15. Februar 2007 17:42 An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] Special SIP scenario Hi all, I want to do a special SIP using and I dont know how to configure it with: originate/bridge/transfer no_media and so on May be it is not possible with freeswitch? Here comes scenario: - one sender device that sends multicast RTP stream to network - one or more receiver devices that only receive RTP data from network The codec is a proprietary one (41), freeswitch can ignore all RTP data. Sender has to get attributs: u=file:/C:/dap/123.wav c=IN IP4 239.194.0.14 t=0 0 m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=recvonly b=AS:128 Receiver has to get attributs: c=IN IP4 239.193.201.69 t=0 0 a=specialExtension:2 a=type:broadcast m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=sendonly Is it possible to use freeswitch API to do this? Many thanks in advance, Thomas Sluschny
The originate syntax allows for multiple calls out with "and" (,) and "or" (|) logic. This should be documented on the wiki for originate syntax. This should also allow you to play a file to the answering party and do key confirmation to initiate a bridge with the original calling party. As far as custom codecs, either for passthrough only, or with transcoding, you would need to make a codec module, at least setup to do the passthrough, the amr and g729 codecs in tree are a good example of this. Typically, you should not ever use codec numbers in the static range (<97 I believe) unless they are officially assigned, that being said, freeswitch will not keep you from writing a module that will work for this. If freeswitch is truly not in the media path at all, you can use no media mode, which would pass through the sdp raw to the other side, in which case you would not even need the passthrough codec, as log of course as the endpoints knew how to handle the multicast. Mike ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas Sent: Wednesday, February 21, 2007 9:02 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Special SIP scenario Hello, unfortunately no one answered til now. So i ask a bit different: 1. Is it possible to create a call to a SIP device where i can define the MessageBody or SDP field on my own (as mentioned in first mail)? I want to INVITE one sender (plays an announcement) and many receivers. These INVITEs should be independent, so if one receiver is out of order all others can be announced anyway. 2. Is it possible to invite a SIP device independend from others? I tested: pacall device1; pacall device2; uuid_bridge sip_uuid1 sip_uuid2 -> this is a little bit sick, i know ;) originate sofia/domain/device1 device2 -> but this works only with 2 devices AND the INVITEs go out successively Has this "media" command anything to do with it? Hope this clears more my problem. Thanks, Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Sluschny, Thomas Gesendet: Donnerstag, 15. Februar 2007 17:42 An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] Special SIP scenario Hi all, I want to do a special SIP using and I dont know how to configure it with: originate/bridge/transfer no_media and so on May be it is not possible with freeswitch? Here comes scenario: - one sender device that sends multicast RTP stream to network - one or more receiver devices that only receive RTP data from network The codec is a proprietary one (41), freeswitch can ignore all RTP data. Sender has to get attributs: u=file:/C:/dap/123.wav <file:///C:\dap\123.wav> c=IN IP4 239.194.0.14 t=0 0 m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=recvonly b=AS:128 Receiver has to get attributs: c=IN IP4 239.193.201.69 t=0 0 a=specialExtension:2 a=type:broadcast m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=sendonly Is it possible to use freeswitch API to do this? Many thanks in advance, Thomas Sluschny
Thank you Mike for the answer. But: i would not only call multiple parties/devices/receivers, but also announce all these receivers with one sender at the same time! To do this the audio data is multiplied in the network, thats why we use (ip-) multicast: see below c=IN IP4 239.194.0.14 And this is also a problem because the "c=" attribute contains normally the own IP and not a multicast one, so: is it possible to set some (all?) SDP attributs directly? (i saw e.g. remote_media_ip in http://wiki.freeswitch.org/wiki/FreeSwitch_Channel_Variables (under Unknown Functionality) Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 21. Februar 2007 15:59 An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] Special SIP scenario The originate syntax allows for multiple calls out with "and" (,) and "or" (|) logic. This should be documented on the wiki for originate syntax. This should also allow you to play a file to the answering party and do key confirmation to initiate a bridge with the original calling party. As far as custom codecs, either for passthrough only, or with transcoding, you would need to make a codec module, at least setup to do the passthrough, the amr and g729 codecs in tree are a good example of this. Typically, you should not ever use codec numbers in the static range (<97 I believe) unless they are officially assigned, that being said, freeswitch will not keep you from writing a module that will work for this. If freeswitch is truly not in the media path at all, you can use no media mode, which would pass through the sdp raw to the other side, in which case you would not even need the passthrough codec, as log of course as the endpoints knew how to handle the multicast. Mike ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas Sent: Wednesday, February 21, 2007 9:02 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Special SIP scenario Hello, unfortunately no one answered til now. So i ask a bit different: 1. Is it possible to create a call to a SIP device where i can define the MessageBody or SDP field on my own (as mentioned in first mail)? I want to INVITE one sender (plays an announcement) and many receivers. These INVITEs should be independent, so if one receiver is out of order all others can be announced anyway. 2. Is it possible to invite a SIP device independend from others? I tested: pacall device1; pacall device2; uuid_bridge sip_uuid1 sip_uuid2 -> this is a little bit sick, i know ;) originate sofia/domain/device1 device2 -> but this works only with 2 devices AND the INVITEs go out successively Has this "media" command anything to do with it? Hope this clears more my problem. Thanks, Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Sluschny, Thomas Gesendet: Donnerstag, 15. Februar 2007 17:42 An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] Special SIP scenario Hi all, I want to do a special SIP using and I dont know how to configure it with: originate/bridge/transfer no_media and so on May be it is not possible with freeswitch? Here comes scenario: - one sender device that sends multicast RTP stream to network - one or more receiver devices that only receive RTP data from network The codec is a proprietary one (41), freeswitch can ignore all RTP data. Sender has to get attributs: u=file:/C:/dap/123.wav <file:///C:\dap\123.wav> c=IN IP4 239.194.0.14 t=0 0 m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=recvonly b=AS:128 Receiver has to get attributs: c=IN IP4 239.193.201.69 t=0 0 a=specialExtension:2 a=type:broadcast m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=sendonly Is it possible to use freeswitch API to do this? Many thanks in advance, Thomas Sluschny
Are you trying to originate the call from freeswitch (requiring freeswitch to actually send the multicast rtp stream) or are you originating from another device that will send the multicast rtp and create the proper sdp, but will send through freeswitch to handle the sending of invites to all the parties? If freeswitch is going to originate the stream, what is the source of the audio they will be sending to all the calls? I assume that you get no return audio from any of the end devices? Mike ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas Sent: Wednesday, February 21, 2007 11:20 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Special SIP scenario Thank you Mike for the answer. But: i would not only call multiple parties/devices/receivers, but also announce all these receivers with one sender at the same time! To do this the audio data is multiplied in the network, thats why we use (ip-) multicast: see below c=IN IP4 239.194.0.14 And this is also a problem because the "c=" attribute contains normally the own IP and not a multicast one, so: is it possible to set some (all?) SDP attributs directly? (i saw e.g. remote_media_ip in http://wiki.freeswitch.org/wiki/FreeSwitch_Channel_Variables (under Unknown Functionality) Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 21. Februar 2007 15:59 An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] Special SIP scenario The originate syntax allows for multiple calls out with "and" (,) and "or" (|) logic. This should be documented on the wiki for originate syntax. This should also allow you to play a file to the answering party and do key confirmation to initiate a bridge with the original calling party. As far as custom codecs, either for passthrough only, or with transcoding, you would need to make a codec module, at least setup to do the passthrough, the amr and g729 codecs in tree are a good example of this. Typically, you should not ever use codec numbers in the static range (<97 I believe) unless they are officially assigned, that being said, freeswitch will not keep you from writing a module that will work for this. If freeswitch is truly not in the media path at all, you can use no media mode, which would pass through the sdp raw to the other side, in which case you would not even need the passthrough codec, as log of course as the endpoints knew how to handle the multicast. Mike ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas Sent: Wednesday, February 21, 2007 9:02 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Special SIP scenario Hello, unfortunately no one answered til now. So i ask a bit different: 1. Is it possible to create a call to a SIP device where i can define the MessageBody or SDP field on my own (as mentioned in first mail)? I want to INVITE one sender (plays an announcement) and many receivers. These INVITEs should be independent, so if one receiver is out of order all others can be announced anyway. 2. Is it possible to invite a SIP device independend from others? I tested: pacall device1; pacall device2; uuid_bridge sip_uuid1 sip_uuid2 -> this is a little bit sick, i know ;) originate sofia/domain/device1 device2 -> but this works only with 2 devices AND the INVITEs go out successively Has this "media" command anything to do with it? Hope this clears more my problem. Thanks, Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Sluschny, Thomas Gesendet: Donnerstag, 15. Februar 2007 17:42 An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] Special SIP scenario Hi all, I want to do a special SIP using and I dont know how to configure it with: originate/bridge/transfer no_media and so on May be it is not possible with freeswitch? Here comes scenario: - one sender device that sends multicast RTP stream to network - one or more receiver devices that only receive RTP data from network The codec is a proprietary one (41), freeswitch can ignore all RTP data. Sender has to get attributs: u=file:/C:/dap/123.wav <file:///C:\dap\123.wav> c=IN IP4 239.194.0.14 t=0 0 m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=recvonly b=AS:128 Receiver has to get attributs: c=IN IP4 239.193.201.69 t=0 0 a=specialExtension:2 a=type:broadcast m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=sendonly Is it possible to use freeswitch API to do this? Many thanks in advance, Thomas Sluschny
There are 1 sender and many receivers (all not freeswitch). If the sender gets an INVITE with the attributs from below (with multicast IP, wav file uri and recvonly) it begins to stream its wav audio data (after ACK). If one receiver gets an INVITE with the attributs from below (with multicast IP, some special attributs and sendonly) it begins to receive an audio stream. All i want is FreeSwitch to send these special INVITEs to the sender end receivers. FreeSwitch has nothing to do with the audio data or format. Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 21. Februar 2007 17:45 An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] Special SIP scenario Are you trying to originate the call from freeswitch (requiring freeswitch to actually send the multicast rtp stream) or are you originating from another device that will send the multicast rtp and create the proper sdp, but will send through freeswitch to handle the sending of invites to all the parties? If freeswitch is going to originate the stream, what is the source of the audio they will be sending to all the calls? I assume that you get no return audio from any of the end devices? Mike ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas Sent: Wednesday, February 21, 2007 11:20 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Special SIP scenario Thank you Mike for the answer. But: i would not only call multiple parties/devices/receivers, but also announce all these receivers with one sender at the same time! To do this the audio data is multiplied in the network, thats why we use (ip-) multicast: see below c=IN IP4 239.194.0.14 And this is also a problem because the "c=" attribute contains normally the own IP and not a multicast one, so: is it possible to set some (all?) SDP attributs directly? (i saw e.g. remote_media_ip in http://wiki.freeswitch.org/wiki/FreeSwitch_Channel_Variables (under Unknown Functionality) Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 21. Februar 2007 15:59 An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] Special SIP scenario The originate syntax allows for multiple calls out with "and" (,) and "or" (|) logic. This should be documented on the wiki for originate syntax. This should also allow you to play a file to the answering party and do key confirmation to initiate a bridge with the original calling party. As far as custom codecs, either for passthrough only, or with transcoding, you would need to make a codec module, at least setup to do the passthrough, the amr and g729 codecs in tree are a good example of this. Typically, you should not ever use codec numbers in the static range (<97 I believe) unless they are officially assigned, that being said, freeswitch will not keep you from writing a module that will work for this. If freeswitch is truly not in the media path at all, you can use no media mode, which would pass through the sdp raw to the other side, in which case you would not even need the passthrough codec, as log of course as the endpoints knew how to handle the multicast. Mike ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas Sent: Wednesday, February 21, 2007 9:02 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Special SIP scenario Hello, unfortunately no one answered til now. So i ask a bit different: 1. Is it possible to create a call to a SIP device where i can define the MessageBody or SDP field on my own (as mentioned in first mail)? I want to INVITE one sender (plays an announcement) and many receivers. These INVITEs should be independent, so if one receiver is out of order all others can be announced anyway. 2. Is it possible to invite a SIP device independend from others? I tested: pacall device1; pacall device2; uuid_bridge sip_uuid1 sip_uuid2 -> this is a little bit sick, i know ;) originate sofia/domain/device1 device2 -> but this works only with 2 devices AND the INVITEs go out successively Has this "media" command anything to do with it? Hope this clears more my problem. Thanks, Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Sluschny, Thomas Gesendet: Donnerstag, 15. Februar 2007 17:42 An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] Special SIP scenario Hi all, I want to do a special SIP using and I dont know how to configure it with: originate/bridge/transfer no_media and so on May be it is not possible with freeswitch? Here comes scenario: - one sender device that sends multicast RTP stream to network - one or more receiver devices that only receive RTP data from network The codec is a proprietary one (41), freeswitch can ignore all RTP data. Sender has to get attributs: u=file:/C:/dap/123.wav <file:///C:\dap\123.wav> c=IN IP4 239.194.0.14 t=0 0 m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=recvonly b=AS:128 Receiver has to get attributs: c=IN IP4 239.193.201.69 t=0 0 a=specialExtension:2 a=type:broadcast m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=sendonly Is it possible to use freeswitch API to do this? Many thanks in advance, Thomas Sluschny
I think that freeswitch with no-media mode enabled would properly handle this today with no modifications. Try it out and let me know. Mike ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas Sent: Wednesday, February 21, 2007 12:11 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Special SIP scenario There are 1 sender and many receivers (all not freeswitch). If the sender gets an INVITE with the attributs from below (with multicast IP, wav file uri and recvonly) it begins to stream its wav audio data (after ACK). If one receiver gets an INVITE with the attributs from below (with multicast IP, some special attributs and sendonly) it begins to receive an audio stream. All i want is FreeSwitch to send these special INVITEs to the sender end receivers. FreeSwitch has nothing to do with the audio data or format. Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 21. Februar 2007 17:45 An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] Special SIP scenario Are you trying to originate the call from freeswitch (requiring freeswitch to actually send the multicast rtp stream) or are you originating from another device that will send the multicast rtp and create the proper sdp, but will send through freeswitch to handle the sending of invites to all the parties? If freeswitch is going to originate the stream, what is the source of the audio they will be sending to all the calls? I assume that you get no return audio from any of the end devices? Mike ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas Sent: Wednesday, February 21, 2007 11:20 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Special SIP scenario Thank you Mike for the answer. But: i would not only call multiple parties/devices/receivers, but also announce all these receivers with one sender at the same time! To do this the audio data is multiplied in the network, thats why we use (ip-) multicast: see below c=IN IP4 239.194.0.14 And this is also a problem because the "c=" attribute contains normally the own IP and not a multicast one, so: is it possible to set some (all?) SDP attributs directly? (i saw e.g. remote_media_ip in http://wiki.freeswitch.org/wiki/FreeSwitch_Channel_Variables (under Unknown Functionality) Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 21. Februar 2007 15:59 An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] Special SIP scenario The originate syntax allows for multiple calls out with "and" (,) and "or" (|) logic. This should be documented on the wiki for originate syntax. This should also allow you to play a file to the answering party and do key confirmation to initiate a bridge with the original calling party. As far as custom codecs, either for passthrough only, or with transcoding, you would need to make a codec module, at least setup to do the passthrough, the amr and g729 codecs in tree are a good example of this. Typically, you should not ever use codec numbers in the static range (<97 I believe) unless they are officially assigned, that being said, freeswitch will not keep you from writing a module that will work for this. If freeswitch is truly not in the media path at all, you can use no media mode, which would pass through the sdp raw to the other side, in which case you would not even need the passthrough codec, as log of course as the endpoints knew how to handle the multicast. Mike ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sluschny, Thomas Sent: Wednesday, February 21, 2007 9:02 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Special SIP scenario Hello, unfortunately no one answered til now. So i ask a bit different: 1. Is it possible to create a call to a SIP device where i can define the MessageBody or SDP field on my own (as mentioned in first mail)? I want to INVITE one sender (plays an announcement) and many receivers. These INVITEs should be independent, so if one receiver is out of order all others can be announced anyway. 2. Is it possible to invite a SIP device independend from others? I tested: pacall device1; pacall device2; uuid_bridge sip_uuid1 sip_uuid2 -> this is a little bit sick, i know ;) originate sofia/domain/device1 device2 -> but this works only with 2 devices AND the INVITEs go out successively Has this "media" command anything to do with it? Hope this clears more my problem. Thanks, Thomas ________________________________ Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Sluschny, Thomas Gesendet: Donnerstag, 15. Februar 2007 17:42 An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] Special SIP scenario Hi all, I want to do a special SIP using and I dont know how to configure it with: originate/bridge/transfer no_media and so on May be it is not possible with freeswitch? Here comes scenario: - one sender device that sends multicast RTP stream to network - one or more receiver devices that only receive RTP data from network The codec is a proprietary one (41), freeswitch can ignore all RTP data. Sender has to get attributs: u=file:/C:/dap/123.wav <file:///C:\dap\123.wav> c=IN IP4 239.194.0.14 t=0 0 m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=recvonly b=AS:128 Receiver has to get attributs: c=IN IP4 239.193.201.69 t=0 0 a=specialExtension:2 a=type:broadcast m=audio 7280 RTP/AVP 41 a=rtpmap:41 PCMA/16000 a=sendonly Is it possible to use freeswitch API to do this? Many thanks in advance, Thomas Sluschny
participants (2)
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Michael Jerris
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Sluschny, Thomas