Hi all, I'm having some trouble with G722 and AudioCodes HD310 in latest 1.2-stable... The phone sends the SDP: v=0 o=1000 946688278 946688278 IN IP4 192.168.0.108 s=- c=IN IP4 192.168.0.108 t=0 0 m=audio 4000 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 But Freeswitch is prepared to accept only G722@8000 as can be seen in: [DEBUG] sofia_glue.c:5181 Audio Codec Compare [G722:9:16000:20:64000]/[G722:9:8000:10:64000] For now I've changed the source to accept 16000 but according to Anthony, as per the RFC, 8000 is the correct value. I've seen some conversations between Nandy Dagondon ans Steven Ayre but I'm not sure if it is the same issue, nonetheless I could not find a solution. has anyone faced a similar problem and found a viable solution? Best, Wesley Akio TuntsCorp.com
It may seem crazy, but you have to use 8000 for g722. If audiocodes is really sending it as 16000 you need to report that to them. http://www.ietf.org/rfc/rfc3551.txt 4.5.2 G722 G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding within 64 kbit/s". The G.722 encoder produces a stream of octets, each of which SHALL be octet-aligned in an RTP packet. The first bit transmitted in the G.722 octet, which is the most significant bit of the higher sub-band sample, SHALL correspond to the most significant bit of the octet in the RTP packet. Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. The octet rate or sample-pair rate is 8,000 Hz. On Jul 8, 2013, at 12:17 PM, Wesley Akio <wesleyakio@tuntscorp.com> wrote:
Hi all,
I'm having some trouble with G722 and AudioCodes HD310 in latest 1.2-stable...
The phone sends the SDP: v=0 o=1000 946688278 946688278 IN IP4 192.168.0.108 s=- c=IN IP4 192.168.0.108 t=0 0 m=audio 4000 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
But Freeswitch is prepared to accept only G722@8000 as can be seen in:
[DEBUG] sofia_glue.c:5181 Audio Codec Compare [G722:9:16000:20:64000]/[G722:9:8000:10:64000]
For now I've changed the source to accept 16000 but according to Anthony, as per the RFC, 8000 is the correct value.
I've seen some conversations between Nandy Dagondon ans Steven Ayre but I'm not sure if it is the same issue, nonetheless I could not find a solution.
has anyone faced a similar problem and found a viable solution?
Yeap, now I know, Anthony explained and pointed me to some literature... :/ "Historical reasons"... I'm playing with it right now and, in light of this information, it seems I missed something really obvious... Audiocodes has both G.722/8000 and G.722/16000 in the codecs list. At first I though it would vary the sample rate but testing it now it seems all it does is change the SDP, since using G.722/8000 results in: 2013-07-08 13:57:13.469291 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16@16000hz 1 channels 20ms Will test a little bit further but it seems to be simpler than anticipated. Wesley Akio TuntsCorp.com On Mon, Jul 8, 2013 at 1:50 PM, Michael Jerris <mike@jerris.com> wrote:
It may seem crazy, but you have to use 8000 for g722. If audiocodes is really sending it as 16000 you need to report that to them.
http://www.ietf.org/rfc/rfc3551.txt
4.5.2 G722
G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding within 64 kbit/s". The G.722 encoder produces a stream of octets, each of which SHALL be octet-aligned in an RTP packet. The first bit transmitted in the G.722 octet, which is the most significant bit of the higher sub-band sample, SHALL correspond to the most significant bit of the octet in the RTP packet.
Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. The octet rate or sample-pair rate is 8,000 Hz.
On Jul 8, 2013, at 12:17 PM, Wesley Akio <wesleyakio@tuntscorp.com> wrote:
Hi all,
I'm having some trouble with G722 and AudioCodes HD310 in latest 1.2-stable...
The phone sends the SDP: v=0 o=1000 946688278 946688278 IN IP4 192.168.0.108 s=- c=IN IP4 192.168.0.108 t=0 0 m=audio 4000 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
But Freeswitch is prepared to accept only G722@8000 as can be seen in:
[DEBUG] sofia_glue.c:5181 Audio Codec Compare [G722:9:16000:20:64000]/[G722:9:8000:10:64000]
For now I've changed the source to accept 16000 but according to Anthony, as per the RFC, 8000 is the correct value.
I've seen some conversations between Nandy Dagondon ans Steven Ayre but I'm not sure if it is the same issue, nonetheless I could not find a solution.
has anyone faced a similar problem and found a viable solution?
_________________________________________________________________________ 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
The 16000 is probably there for interop with other devices that are incorrect then, 8000 is the one you should use with FS. -Steve On 8 July 2013 18:04, Wesley Akio <wesleyakio@tuntscorp.com> wrote:
Yeap, now I know, Anthony explained and pointed me to some literature... :/ "Historical reasons"...
I'm playing with it right now and, in light of this information, it seems I missed something really obvious...
Audiocodes has both G.722/8000 and G.722/16000 in the codecs list. At first I though it would vary the sample rate but testing it now it seems all it does is change the SDP, since using G.722/8000 results in: 2013-07-08 13:57:13.469291 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16@16000hz 1 channels 20ms
Will test a little bit further but it seems to be simpler than anticipated.
Wesley Akio TuntsCorp.com
On Mon, Jul 8, 2013 at 1:50 PM, Michael Jerris <mike@jerris.com> wrote:
It may seem crazy, but you have to use 8000 for g722. If audiocodes is really sending it as 16000 you need to report that to them.
http://www.ietf.org/rfc/rfc3551.txt
4.5.2 G722
G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding within 64 kbit/s". The G.722 encoder produces a stream of octets, each of which SHALL be octet-aligned in an RTP packet. The first bit transmitted in the G.722 octet, which is the most significant bit of the higher sub-band sample, SHALL correspond to the most significant bit of the octet in the RTP packet.
Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. The octet rate or sample-pair rate is 8,000 Hz.
On Jul 8, 2013, at 12:17 PM, Wesley Akio <wesleyakio@tuntscorp.com> wrote:
Hi all,
I'm having some trouble with G722 and AudioCodes HD310 in latest 1.2-stable...
The phone sends the SDP: v=0 o=1000 946688278 946688278 IN IP4 192.168.0.108 s=- c=IN IP4 192.168.0.108 t=0 0 m=audio 4000 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
But Freeswitch is prepared to accept only G722@8000 as can be seen in:
[DEBUG] sofia_glue.c:5181 Audio Codec Compare [G722:9:16000:20:64000]/[G722:9:8000:10:64000]
For now I've changed the source to accept 16000 but according to Anthony, as per the RFC, 8000 is the correct value.
I've seen some conversations between Nandy Dagondon ans Steven Ayre but I'm not sure if it is the same issue, nonetheless I could not find a solution.
has anyone faced a similar problem and found a viable solution?
_________________________________________________________________________ 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
Hi Wesley, I still have this issue on the sideline. You mentioned you "changed the source". Can you share the changes made? Thanks. /Nandy On Tue, Jul 9, 2013 at 3:22 AM, Steven Ayre <steveayre@gmail.com> wrote:
The 16000 is probably there for interop with other devices that are incorrect then, 8000 is the one you should use with FS.
-Steve
On 8 July 2013 18:04, Wesley Akio <wesleyakio@tuntscorp.com> wrote:
Yeap, now I know, Anthony explained and pointed me to some literature... :/ "Historical reasons"...
I'm playing with it right now and, in light of this information, it seems I missed something really obvious...
Audiocodes has both G.722/8000 and G.722/16000 in the codecs list. At first I though it would vary the sample rate but testing it now it seems all it does is change the SDP, since using G.722/8000 results in: 2013-07-08 13:57:13.469291 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16@16000hz 1 channels 20ms
Will test a little bit further but it seems to be simpler than anticipated.
Wesley Akio TuntsCorp.com
On Mon, Jul 8, 2013 at 1:50 PM, Michael Jerris <mike@jerris.com> wrote:
It may seem crazy, but you have to use 8000 for g722. If audiocodes is really sending it as 16000 you need to report that to them.
http://www.ietf.org/rfc/rfc3551.txt
4.5.2 G722
G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding within 64 kbit/s". The G.722 encoder produces a stream of octets, each of which SHALL be octet-aligned in an RTP packet. The first bit transmitted in the G.722 octet, which is the most significant bit of the higher sub-band sample, SHALL correspond to the most significant bit of the octet in the RTP packet.
Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. The octet rate or sample-pair rate is 8,000 Hz.
On Jul 8, 2013, at 12:17 PM, Wesley Akio <wesleyakio@tuntscorp.com> wrote:
Hi all,
I'm having some trouble with G722 and AudioCodes HD310 in latest 1.2-stable...
The phone sends the SDP: v=0 o=1000 946688278 946688278 IN IP4 192.168.0.108 s=- c=IN IP4 192.168.0.108 t=0 0 m=audio 4000 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
But Freeswitch is prepared to accept only G722@8000 as can be seen in:
[DEBUG] sofia_glue.c:5181 Audio Codec Compare [G722:9:16000:20:64000]/[G722:9:8000:10:64000]
For now I've changed the source to accept 16000 but according to Anthony, as per the RFC, 8000 is the correct value.
I've seen some conversations between Nandy Dagondon ans Steven Ayre but I'm not sure if it is the same issue, nonetheless I could not find a solution.
has anyone faced a similar problem and found a viable solution?
_________________________________________________________________________ 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
_________________________________________________________________________ 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
participants (4)
-
Michael Jerris
-
Nandy Dagondon
-
Steven Ayre
-
Wesley Akio