Hi all! I’m trying to connect my FS to a Sonus SIP trunk. I followed the instruction at http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus but it did not work. At the beginning of a call voice quality is good. After a while it changes to choppy. I don’t know if it’s the same problem: When I call the Tetris extension via Sonus SIP trunk the sound is too fast and I’m getting log entries like the following one: [...] 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME not supported, changing our end from 20 to 10 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from PCMA@20ms@8000hz to PCMA@10ms@8000hz 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global timer resolution to 10ms to handle interval 10 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer [soft] 80 bytes per 10ms 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write Buffer at 160 bytes to accommodate 320->160 [...] This problem is fixed by adding the following line to conf/sip_profiles/external.xml: <param name="rtp-autofix-timing" value="false"/> Any hints? /micha
Sonus sucks. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 26/07/2011 à 16:52, michael knop a écrit :
Hi all!
I’m trying to connect my FS to a Sonus SIP trunk. I followed the instruction at
http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
but it did not work. At the beginning of a call voice quality is good. After a while it changes to choppy.
I don’t know if it’s the same problem: When I call the Tetris extension via Sonus SIP trunk the sound is too fast and I’m getting log entries like the following one:
[...] 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME not supported, changing our end from 20 to 10 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from PCMA@20ms@8000hz to PCMA@10ms@8000hz 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global timer resolution to 10ms to handle interval 10 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer [soft] 80 bytes per 10ms 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write Buffer at 160 bytes to accommodate 320->160 [...]
This problem is fixed by adding the following line to conf/sip_profiles/external.xml:
<param name="rtp-autofix-timing" value="false"/>
Any hints?
/micha
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
full ack Am 27.07.2011 11:51, schrieb David Ponzone:
Sonus sucks.
David Ponzone Direction Technique email: david.ponzone@ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34
Service ClientIPeva tel: 0811 46 26 26 www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>- www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
/Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. /*/IPeva/*/décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur./ / /
Le 26/07/2011 à 16:52, michael knop a écrit :
Hi all!
I’m trying to connect my FS to a Sonus SIP trunk. I followed the instruction at
http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
but it did not work. At the beginning of a call voice quality is good. After a while it changes to choppy.
I don’t know if it’s the same problem: When I call the Tetris extension via Sonus SIP trunk the sound is too fast and I’m getting log entries like the following one:
[...] 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME not supported, changing our end from 20 to 10 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from PCMA@20ms@8000hz to PCMA@10ms@8000hz 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global timer resolution to 10ms to handle interval 10 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer [soft] 80 bytes per 10ms 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write Buffer at 160 bytes to accommodate 320->160 [...]
This problem is fixed by adding the following line to conf/sip_profiles/external.xml:
<param name="rtp-autofix-timing" value="false"/>
Any hints?
/micha
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
Hello Micha, How much time does it take until the sound is choppy? We have been connected to Sonus provider up to a week ago. Incoming calls started being choppy after about 15 minutes (outgoing calls were ok). We also had inconsistent problems with DTMF. We ended it by changing a supplier... __Yehavi: 2011/7/26 michael knop <michael.knop@hcu-hamburg.de>
Hi all!
I’m trying to connect my FS to a Sonus SIP trunk. I followed the instruction at
http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
but it did not work. At the beginning of a call voice quality is good. After a while it changes to choppy.
I don’t know if it’s the same problem: When I call the Tetris extension via Sonus SIP trunk the sound is too fast and I’m getting log entries like the following one:
[...] 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME not supported, changing our end from 20 to 10 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from PCMA@20ms@8000hz to PCMA@10ms@8000hz 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global timer resolution to 10ms to handle interval 10 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer [soft] 80 bytes per 10ms 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write Buffer at 160 bytes to accommodate 320->160 [...]
This problem is fixed by adding the following line to conf/sip_profiles/external.xml:
<param name="rtp-autofix-timing" value="false"/>
Any hints?
/micha
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
Yehavi, unfortunately changing the provider is not an option. If I can’t get FreeSWITCH to work with Sonus I can’t switch to FreeSWITCH. Then if have to stay using Asterisk. To me it looks like that its nondeterministic how much time it takes until the sound is choppy. Sometimes sound quality is brilliant for several minutes. Sometimes the call starts with choppy sound. /micha Am 27.07.2011 20:54, schrieb Yehavi Bourvine:
Hello Micha, How much time does it take until the sound is choppy? We have been connected to Sonus provider up to a week ago. Incoming calls started being choppy after about 15 minutes (outgoing calls were ok). We also had inconsistent problems with DTMF. We ended it by changing a supplier... __Yehavi:
2011/7/26 michael knop <michael.knop@hcu-hamburg.de <mailto:michael.knop@hcu-hamburg.de>>
Hi all!
I’m trying to connect my FS to a Sonus SIP trunk. I followed the instruction at
http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
but it did not work. At the beginning of a call voice quality is good. After a while it changes to choppy.
I don’t know if it’s the same problem: When I call the Tetris extension via Sonus SIP trunk the sound is too fast and I’m getting log entries like the following one:
[...] 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME not supported, changing our end from 20 to 10 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from PCMA@20ms@8000hz to PCMA@10ms@8000hz 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global timer resolution to 10ms to handle interval 10 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer [soft] 80 bytes per 10ms 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write Buffer at 160 bytes to accommodate 320->160 [...]
This problem is fixed by adding the following line to conf/sip_profiles/external.xml:
<param name="rtp-autofix-timing" value="false"/>
Any hints?
/micha
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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
Update: Call starts with good sound quality. After the following log message sound is choppy: 2011-07-28 12:07:22.639151 [DEBUG] sofia.c:5094 Duplicate SDP v=0 o=Sonus_UAC 8739 8900 IN IP4 XXX.XXX.XXX.XXX s=SIP Media Capabilities c=IN IP4 YYY.YYY.YYY.YYY t=0 0 m=audio 20320 RTP/AVP 8 0 18 100 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=maxptime:10 /micha Am 26.07.2011 16:52, schrieb michael knop:
Hi all!
I’m trying to connect my FS to a Sonus SIP trunk. I followed the instruction at
http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
but it did not work. At the beginning of a call voice quality is good. After a while it changes to choppy.
I don’t know if it’s the same problem: When I call the Tetris extension via Sonus SIP trunk the sound is too fast and I’m getting log entries like the following one:
[...] 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME not supported, changing our end from 20 to 10 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from PCMA@20ms@8000hz to PCMA@10ms@8000hz 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global timer resolution to 10ms to handle interval 10 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer [soft] 80 bytes per 10ms 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write Buffer at 160 bytes to accommodate 320->160 [...]
This problem is fixed by adding the following line to conf/sip_profiles/external.xml:
<param name="rtp-autofix-timing" value="false"/>
Any hints?
/micha
Hi all! I'm not sure, if my problem is caused by Sonus or is it's just a problem while negotiating the audio params. I put the log into pastebin: <http://pastebin.freeswitch.org/16913> Call starts with good sound quality. After "Duplicate SDP" (row 261) sound is choppy. /micha
On Fri, Jul 29, 2011 at 3:58 AM, michael knop <michael.knop@hcu-hamburg.de> wrote:
Hi all!
I'm not sure, if my problem is caused by Sonus or is it's just a problem while negotiating the audio params. I put the log into pastebin: <http://pastebin.freeswitch.org/16913>
Call starts with good sound quality. After "Duplicate SDP" (row 261) sound is choppy.
I know Anthony was recently doing more working to play nice with the brokenness that is Sonus; you might want to check out the updates and see if they make things better. I don't recall for sure what it was he was addressing, but it might be worth a shot. Gabe
Have you thought of opening a Jira? /b On Jul 28, 2011, at 5:14 AM, michael knop wrote:
Update:
Call starts with good sound quality. After the following log message sound is choppy:
2011-07-28 12:07:22.639151 [DEBUG] sofia.c:5094 Duplicate SDP v=0 o=Sonus_UAC 8739 8900 IN IP4 XXX.XXX.XXX.XXX s=SIP Media Capabilities c=IN IP4 YYY.YYY.YYY.YYY t=0 0 m=audio 20320 RTP/AVP 8 0 18 100 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=maxptime:10
/micha
Am 26.07.2011 16:52, schrieb michael knop:
Hi all!
I’m trying to connect my FS to a Sonus SIP trunk. I followed the instruction at
http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
but it did not work. At the beginning of a call voice quality is good. After a while it changes to choppy.
I don’t know if it’s the same problem: When I call the Tetris extension via Sonus SIP trunk the sound is too fast and I’m getting log entries like the following one:
[...] 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME not supported, changing our end from 20 to 10 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from PCMA@20ms@8000hz to PCMA@10ms@8000hz 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global timer resolution to 10ms to handle interval 10 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer [soft] 80 bytes per 10ms 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write Buffer at 160 bytes to accommodate 320->160 [...]
This problem is fixed by adding the following line to conf/sip_profiles/external.xml:
<param name="rtp-autofix-timing" value="false"/>
Any hints?
/micha
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
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 (5)
-
Brian West
-
David Ponzone
-
Gabriel Gunderson
-
michael knop
-
Yehavi Bourvine