suggest best codecs to use with freeswitch for good voice quality ??
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine suggest best codecs to use with freeswitch and asterisk for good voice quality ??
Rather than using asterisk + chan_dongle, perhaps look at mod_gsmopen? It'll be a simpler setup. On 3 April 2015 at 19:50, Shabbir abbasi <shabbirabbasi92@gmail.com> wrote:
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine
suggest best codecs to use with freeswitch and asterisk for good voice quality ??
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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i have tried mod_gsmopen it has many problems, and at same machine chan_dongle with asterisk working fine, mod_gsmopen is marking device as dead after some time and don't try again to connect to device untill you manualy issue command "gsm reload" or reload mod_gsmopen and some time it says that module is in use and shutdown whole freeswitch,i spent whole day but it is not working fine like chan_dongle, On Sat, Apr 4, 2015 at 3:04 AM, Steven Ayre <steveayre@gmail.com> wrote:
Rather than using asterisk + chan_dongle, perhaps look at mod_gsmopen? It'll be a simpler setup.
On 3 April 2015 at 19:50, Shabbir abbasi <shabbirabbasi92@gmail.com> wrote:
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine
suggest best codecs to use with freeswitch and asterisk for good voice quality ??
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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probably the best you can do is limit the number of transcodings and/or resamplings skype uses silk freeswitch core uses L16 sip session uses <your choice> asterisk core uses ? audio is presented to the dongle as ? cellular network uses gsm if ? = L16 then make the sip session use L16 use the loopback interface so that MTU is not an issue On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi <shabbirabbasi92@gmail.com> wrote:
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine
suggest best codecs to use with freeswitch and asterisk for good voice quality ??
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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thank you for reply it is freeswitch.conf <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h ,G722,PCMU,PCMA,GSM"/> <profile name="freeswitch-sip"> <gateways> <gateway name="asterisk-local"> <param name="proxy" value="127.0.0.1:5060"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways> <domains> <domain name="all" alias="true" parse="false"/> </domains> <settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="false"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="asterisk"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5050"/> <param name="dialplan" value="XML"/> <param name="dtmf-type" value="info"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="vad" value="none"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <param name="ext-rtp-ip" value="127.0.0.1"/> <param name="ext-sip-ip" value="127.0.0.1"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="auth-all-packets" value="false"/> <param name="challenge-realm" value="auto_from"/> </settings> and here is asterisk sip.con disallow=all allow=ulaw allow=alaw what i need to change ?? On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
probably the best you can do is limit the number of transcodings and/or resamplings
skype uses silk freeswitch core uses L16 sip session uses <your choice> asterisk core uses ? audio is presented to the dongle as ? cellular network uses gsm
if ? = L16 then make the sip session use L16 use the loopback interface so that MTU is not an issue
On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi <shabbirabbasi92@gmail.com> wrote:
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine
suggest best codecs to use with freeswitch and asterisk for good voice quality ??
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/> for the asterisk side you will have to figure that out yourself. On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi <shabbirabbasi92@gmail.com> wrote:
thank you for reply it is freeswitch.conf <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<profile name="freeswitch-sip"> <gateways> <gateway name="asterisk-local"> <param name="proxy" value="127.0.0.1:5060"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways>
<domains> <domain name="all" alias="true" parse="false"/> </domains>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="false"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="asterisk"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5050"/> <param name="dialplan" value="XML"/> <param name="dtmf-type" value="info"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="vad" value="none"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <param name="ext-rtp-ip" value="127.0.0.1"/> <param name="ext-sip-ip" value="127.0.0.1"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="auth-all-packets" value="false"/> <param name="challenge-realm" value="auto_from"/> </settings>
and here is asterisk sip.con disallow=all allow=ulaw allow=alaw
what i need to change ??
On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven <rtreleaven@bunnykick.ca
wrote:
probably the best you can do is limit the number of transcodings and/or resamplings
skype uses silk freeswitch core uses L16 sip session uses <your choice> asterisk core uses ? audio is presented to the dongle as ? cellular network uses gsm
if ? = L16 then make the sip session use L16 use the loopback interface so that MTU is not an issue
On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi <shabbirabbasi92@gmail.com
wrote:
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine
suggest best codecs to use with freeswitch and asterisk for good voice quality ??
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Since one of the legs is GSM the best you can hope for is slightly worse than GSM because you are transcoding. On Sat, Apr 4, 2015 at 2:01 PM, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/>
for the asterisk side you will have to figure that out yourself.
On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi <shabbirabbasi92@gmail.com
wrote:
thank you for reply it is freeswitch.conf <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<profile name="freeswitch-sip"> <gateways> <gateway name="asterisk-local"> <param name="proxy" value="127.0.0.1:5060"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways>
<domains> <domain name="all" alias="true" parse="false"/> </domains>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="false"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="asterisk"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5050"/> <param name="dialplan" value="XML"/> <param name="dtmf-type" value="info"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="vad" value="none"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <param name="ext-rtp-ip" value="127.0.0.1"/> <param name="ext-sip-ip" value="127.0.0.1"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="auth-all-packets" value="false"/> <param name="challenge-realm" value="auto_from"/> </settings>
and here is asterisk sip.con disallow=all allow=ulaw allow=alaw
what i need to change ??
On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
probably the best you can do is limit the number of transcodings and/or resamplings
skype uses silk freeswitch core uses L16 sip session uses <your choice> asterisk core uses ? audio is presented to the dongle as ? cellular network uses gsm
if ? = L16 then make the sip session use L16 use the loopback interface so that MTU is not an issue
On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine
suggest best codecs to use with freeswitch and asterisk for good voice quality ??
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
That's not necessarily the case - GSM is not the only codec used by mobile networks, there's others such as AMR On 4 April 2015 at 19:11, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
Since one of the legs is GSM the best you can hope for is slightly worse than GSM because you are transcoding.
On Sat, Apr 4, 2015 at 2:01 PM, Russell Treleaven <rtreleaven@bunnykick.ca
wrote:
for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/>
for the asterisk side you will have to figure that out yourself.
On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
thank you for reply it is freeswitch.conf <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<profile name="freeswitch-sip"> <gateways> <gateway name="asterisk-local"> <param name="proxy" value="127.0.0.1:5060"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways>
<domains> <domain name="all" alias="true" parse="false"/> </domains>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="false"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="asterisk"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5050"/> <param name="dialplan" value="XML"/> <param name="dtmf-type" value="info"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="vad" value="none"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <param name="ext-rtp-ip" value="127.0.0.1"/> <param name="ext-sip-ip" value="127.0.0.1"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="auth-all-packets" value="false"/> <param name="challenge-realm" value="auto_from"/> </settings>
and here is asterisk sip.con disallow=all allow=ulaw allow=alaw
what i need to change ??
On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
probably the best you can do is limit the number of transcodings and/or resamplings
skype uses silk freeswitch core uses L16 sip session uses <your choice> asterisk core uses ? audio is presented to the dongle as ? cellular network uses gsm
if ? = L16 then make the sip session use L16 use the loopback interface so that MTU is not an issue
On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine
suggest best codecs to use with freeswitch and asterisk for good voice quality ??
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
I think that AMR is one of the codecs that does not allow for transcoding, at least with the in-tree module. Might require something additional. -- Yossi Neiman On 04/07/2015 04:57 AM, Steven Ayre wrote:
That's not necessarily the case - GSM is not the only codec used by mobile networks, there's others such as AMR
On 4 April 2015 at 19:11, Russell Treleaven <rtreleaven@bunnykick.ca <mailto:rtreleaven@bunnykick.ca>> wrote:
Since one of the legs is GSM the best you can hope for is slightly worse than GSM because you are transcoding.
On Sat, Apr 4, 2015 at 2:01 PM, Russell Treleaven <rtreleaven@bunnykick.ca <mailto:rtreleaven@bunnykick.ca>> wrote:
for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/>
for the asterisk side you will have to figure that out yourself.
On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi <shabbirabbasi92@gmail.com <mailto:shabbirabbasi92@gmail.com>> wrote:
thank you for reply it is freeswitch.conf <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<profile name="freeswitch-sip"> <gateways> <gateway name="asterisk-local"> <param name="proxy" value="127.0.0.1:5060 <http://127.0.0.1:5060>"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways>
<domains> <domain name="all" alias="true" parse="false"/> </domains>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="false"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="asterisk"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5050"/> <param name="dialplan" value="XML"/> <param name="dtmf-type" value="info"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="vad" value="none"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <param name="ext-rtp-ip" value="127.0.0.1"/> <param name="ext-sip-ip" value="127.0.0.1"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="auth-all-packets" value="false"/> <param name="challenge-realm" value="auto_from"/> </settings>
and here is asterisk sip.con disallow=all allow=ulaw allow=alaw
what i need to change ??
On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven <rtreleaven@bunnykick.ca <mailto:rtreleaven@bunnykick.ca>> wrote:
probably the best you can do is limit the number of transcodings and/or resamplings
skype uses silk freeswitch core uses L16 sip session uses <your choice> asterisk core uses ? audio is presented to the dongle as ? cellular network uses gsm
if ? = L16 then make the sip session use L16 use the loopback interface so that MTU is not an issue
On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi <shabbirabbasi92@gmail.com <mailto:shabbirabbasi92@gmail.com>> wrote:
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine
suggest best codecs to use with freeswitch and asterisk for good voice quality ??
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Can the GSM dongle do AMR? On Tue, Apr 7, 2015 at 12:42 PM, I put the Who? in Mishehu < mishehu@freeswitch.org> wrote:
I think that AMR is one of the codecs that does not allow for transcoding, at least with the in-tree module. Might require something additional.
-- Yossi Neiman
On 04/07/2015 04:57 AM, Steven Ayre wrote:
That's not necessarily the case - GSM is not the only codec used by mobile networks, there's others such as AMR
On 4 April 2015 at 19:11, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
Since one of the legs is GSM the best you can hope for is slightly worse than GSM because you are transcoding.
On Sat, Apr 4, 2015 at 2:01 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/>
for the asterisk side you will have to figure that out yourself.
On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
thank you for reply it is freeswitch.conf <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<profile name="freeswitch-sip"> <gateways> <gateway name="asterisk-local"> <param name="proxy" value="127.0.0.1:5060"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways>
<domains> <domain name="all" alias="true" parse="false"/> </domains>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="false"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="asterisk"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5050"/> <param name="dialplan" value="XML"/> <param name="dtmf-type" value="info"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="vad" value="none"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <param name="ext-rtp-ip" value="127.0.0.1"/> <param name="ext-sip-ip" value="127.0.0.1"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="auth-all-packets" value="false"/> <param name="challenge-realm" value="auto_from"/> </settings>
and here is asterisk sip.con disallow=all allow=ulaw allow=alaw
what i need to change ??
On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
probably the best you can do is limit the number of transcodings and/or resamplings
skype uses silk freeswitch core uses L16 sip session uses <your choice> asterisk core uses ? audio is presented to the dongle as ? cellular network uses gsm
if ? = L16 then make the sip session use L16 use the loopback interface so that MTU is not an issue
On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine
suggest best codecs to use with freeswitch and asterisk for good voice quality ??
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Yes, GSM dongle can do AMR, but the I/O (with PC) in L16. So, the best codec, in this case, will be L16. 2015-04-07 19:59 GMT+03:00 Russell Treleaven <rtreleaven@bunnykick.ca>:
Can the GSM dongle do AMR?
On Tue, Apr 7, 2015 at 12:42 PM, I put the Who? in Mishehu < mishehu@freeswitch.org> wrote:
I think that AMR is one of the codecs that does not allow for transcoding, at least with the in-tree module. Might require something additional.
-- Yossi Neiman
On 04/07/2015 04:57 AM, Steven Ayre wrote:
That's not necessarily the case - GSM is not the only codec used by mobile networks, there's others such as AMR
On 4 April 2015 at 19:11, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
Since one of the legs is GSM the best you can hope for is slightly worse than GSM because you are transcoding.
On Sat, Apr 4, 2015 at 2:01 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/>
for the asterisk side you will have to figure that out yourself.
On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
thank you for reply it is freeswitch.conf <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<profile name="freeswitch-sip"> <gateways> <gateway name="asterisk-local"> <param name="proxy" value="127.0.0.1:5060"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways>
<domains> <domain name="all" alias="true" parse="false"/> </domains>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="false"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="asterisk"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5050"/> <param name="dialplan" value="XML"/> <param name="dtmf-type" value="info"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="vad" value="none"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <param name="ext-rtp-ip" value="127.0.0.1"/> <param name="ext-sip-ip" value="127.0.0.1"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="auth-all-packets" value="false"/> <param name="challenge-realm" value="auto_from"/> </settings>
and here is asterisk sip.con disallow=all allow=ulaw allow=alaw
what i need to change ??
On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
probably the best you can do is limit the number of transcodings and/or resamplings
skype uses silk freeswitch core uses L16 sip session uses <your choice> asterisk core uses ? audio is presented to the dongle as ? cellular network uses gsm
if ? = L16 then make the sip session use L16 use the loopback interface so that MTU is not an issue
On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
> for this setup > skype --> freeswitch(mod_skypopen --> mod_sofia) ---> > asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine > > suggest best codecs to use with freeswitch and asterisk for good > voice quality ?? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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
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dmitry lysenko wrote "Yes, GSM dongle can do AMR, but the I/O (with PC) in L16. So, the best codec, in this case, will be L16." this mean i try rtreleaven's suggestion ?? "<rtreleaven@bunnykick.ca> wrote: for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/>" ? On Tue, Apr 7, 2015 at 11:33 PM, Dmitry Lysenko <dvl36.ripe.nick@gmail.com> wrote:
Yes, GSM dongle can do AMR, but the I/O (with PC) in L16. So, the best codec, in this case, will be L16.
2015-04-07 19:59 GMT+03:00 Russell Treleaven <rtreleaven@bunnykick.ca>:
Can the GSM dongle do AMR?
On Tue, Apr 7, 2015 at 12:42 PM, I put the Who? in Mishehu < mishehu@freeswitch.org> wrote:
I think that AMR is one of the codecs that does not allow for transcoding, at least with the in-tree module. Might require something additional.
-- Yossi Neiman
On 04/07/2015 04:57 AM, Steven Ayre wrote:
That's not necessarily the case - GSM is not the only codec used by mobile networks, there's others such as AMR
On 4 April 2015 at 19:11, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
Since one of the legs is GSM the best you can hope for is slightly worse than GSM because you are transcoding.
On Sat, Apr 4, 2015 at 2:01 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/>
for the asterisk side you will have to figure that out yourself.
On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
thank you for reply it is freeswitch.conf <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<profile name="freeswitch-sip"> <gateways> <gateway name="asterisk-local"> <param name="proxy" value="127.0.0.1:5060"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways>
<domains> <domain name="all" alias="true" parse="false"/> </domains>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="false"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="asterisk"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5050"/> <param name="dialplan" value="XML"/> <param name="dtmf-type" value="info"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="vad" value="none"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <param name="ext-rtp-ip" value="127.0.0.1"/> <param name="ext-sip-ip" value="127.0.0.1"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="auth-all-packets" value="false"/> <param name="challenge-realm" value="auto_from"/> </settings>
and here is asterisk sip.con disallow=all allow=ulaw allow=alaw
what i need to change ??
On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
> probably the best you can do is limit the number of transcodings > and/or resamplings > > skype uses silk > freeswitch core uses L16 > sip session uses <your choice> > asterisk core uses ? > audio is presented to the dongle as ? > cellular network uses gsm > > if ? = L16 then make the sip session use L16 > use the loopback interface so that MTU is not an issue > > > On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi < > shabbirabbasi92@gmail.com> wrote: > >> for this setup >> skype --> freeswitch(mod_skypopen --> mod_sofia) ---> >> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine >> >> suggest best codecs to use with freeswitch and asterisk for good >> voice quality ?? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.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 > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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 >
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PCM/L16 sample rate of GSM dongle is 8000. So, <action application="set" data="absolute_codec_string=L16@8000h"/>" Best wishes, Dmitry. 2015-04-08 7:19 GMT+03:00 Shabbir abbasi <shabbirabbasi92@gmail.com>:
dmitry lysenko wrote "Yes, GSM dongle can do AMR, but the I/O (with PC) in L16. So, the best codec, in this case, will be L16." this mean i try rtreleaven's suggestion ?? "<rtreleaven@bunnykick.ca> wrote: for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/>"
?
On Tue, Apr 7, 2015 at 11:33 PM, Dmitry Lysenko <dvl36.ripe.nick@gmail.com
wrote:
Yes, GSM dongle can do AMR, but the I/O (with PC) in L16. So, the best codec, in this case, will be L16.
2015-04-07 19:59 GMT+03:00 Russell Treleaven <rtreleaven@bunnykick.ca>:
Can the GSM dongle do AMR?
On Tue, Apr 7, 2015 at 12:42 PM, I put the Who? in Mishehu < mishehu@freeswitch.org> wrote:
I think that AMR is one of the codecs that does not allow for transcoding, at least with the in-tree module. Might require something additional.
-- Yossi Neiman
On 04/07/2015 04:57 AM, Steven Ayre wrote:
That's not necessarily the case - GSM is not the only codec used by mobile networks, there's others such as AMR
On 4 April 2015 at 19:11, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
Since one of the legs is GSM the best you can hope for is slightly worse than GSM because you are transcoding.
On Sat, Apr 4, 2015 at 2:01 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/>
for the asterisk side you will have to figure that out yourself.
On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
> thank you for reply > it is freeswitch.conf > <X-PRE-PROCESS cmd="set" > data="sound_prefix=$${sounds_dir}/en/us/callie"/> > <X-PRE-PROCESS cmd="set" data="default_country=US"/> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h > ,G7221@16000h,G722,PCMU,PCMA,GSM"/> > > <profile name="freeswitch-sip"> > <gateways> > <gateway name="asterisk-local"> > <param name="proxy" value="127.0.0.1:5060"/> > <param name="retry-seconds" value="30"/> > <param name="caller-id-in-from" value="true"/> > </gateway> > </gateways> > > <domains> > <domain name="all" alias="true" parse="false"/> > </domains> > > <settings> > <param name="debug" value="1"/> > <param name="sip-trace" value="no"/> > <param name="log-auth-failures" value="false"/> > <param name="forward-unsolicited-mwi-notify" > value="false"/> > <param name="context" value="asterisk"/> > <param name="rfc2833-pt" value="101"/> > <param name="sip-port" value="5050"/> > <param name="dialplan" value="XML"/> > <param name="dtmf-type" value="info"/> > <param name="inbound-codec-prefs" > value="$${global_codec_prefs}"/> > <param name="outbound-codec-prefs" > value="$${global_codec_prefs}"/> > <param name="use-rtp-timer" value="true"/> > <param name="rtp-timer-name" value="soft"/> > <param name="rtp-timeout-sec" value="300"/> > <param name="rtp-hold-timeout-sec" value="1800"/> > <param name="vad" value="none"/> > <param name="rtp-ip" value="127.0.0.1"/> > <param name="sip-ip" value="127.0.0.1"/> > <param name="ext-rtp-ip" value="127.0.0.1"/> > <param name="ext-sip-ip" value="127.0.0.1"/> > <param name="inbound-codec-negotiation" > value="generous"/> > <param name="tls" value="false"/> > <param name="nonce-ttl" value="60"/> > <param name="auth-calls" value="false"/> > <param name="auth-all-packets" value="false"/> > <param name="challenge-realm" value="auto_from"/> > </settings> > > > and here is asterisk sip.con > disallow=all > allow=ulaw > allow=alaw > > what i need to change ?? > > On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven < > rtreleaven@bunnykick.ca> wrote: > >> probably the best you can do is limit the number of transcodings >> and/or resamplings >> >> skype uses silk >> freeswitch core uses L16 >> sip session uses <your choice> >> asterisk core uses ? >> audio is presented to the dongle as ? >> cellular network uses gsm >> >> if ? = L16 then make the sip session use L16 >> use the loopback interface so that MTU is not an issue >> >> >> On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi < >> shabbirabbasi92@gmail.com> wrote: >> >>> for this setup >>> skype --> freeswitch(mod_skypopen --> mod_sofia) ---> >>> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine >>> >>> suggest best codecs to use with freeswitch and asterisk for good >>> voice quality ?? >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.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 >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.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 > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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 >
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participants (5)
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Dmitry Lysenko
-
I put the Who? in Mishehu
-
Russell Treleaven
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Shabbir abbasi
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Steven Ayre