Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)
Hello, when calling from Fritzbox to a Snom Phone , sound is fine. But when calling an internal Freeswitch number (conference, mailbox) i hear a very choppy voice coming from the fritzbox side. I think it may have to do with the ptime 20msec/30msec. Example: When calling from the fritzbox to a voicemail then the annoucement from Freeswitch is choppy (too slow with interrups), but the recorded message is fine. Did anybody experience the same problem? Best regards Peter Here are some SIP messages: U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060 INVITE sip:0123456@my.domain SIP/2.0. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>. Call-ID: 3125316C8A2A13B8@112.xxx.xx.xxx. CSeq: 55 INVITE. Contact: <sip:02xxxxxxxxx@112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>. Max-Forwards: 70. Expires: 120. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Supported: 100rel,replaces,timer. Allow-Events: telephone-event,refer. Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH. Content-Type: application/sdp. Accept: application/sdp, multipart/mixed. Accept-Encoding: identity. Content-Length: 359. . v=0. o=user 16423733 16423733 IN IP4 112.xxx.xx.xxx. s=call. c=IN IP4 112.xxx.xx.xxx. t=0 0. m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101. a=sendrecv. a=rtpmap:2 G726-32/8000. a=rtpmap:102 G726-32/8000. a=rtpmap:100 G726-40/8000. a=rtpmap:99 G726-24/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=rtcp:7079. # U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>. Call-ID: 3125316C8A2A13B8@112.xxx.xx.xxx. CSeq: 55 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Content-Length: 0. . # U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>;tag=7t1e8BQg5B7yK. Call-ID: 3125316C8A2A13B8@112.xxx.xx.xxx. CSeq: 55 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Proxy-Authenticate: Digest realm="my.domain", nonce="900b46a0-8d88-11de-a6a1-098738f35adb", algorithm=MD5, qop="auth". Content-Length: 0. . # U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060 ACK sip:0123456@my.domain SIP/2.0. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>;tag=7t1e8BQg5B7yK. Call-ID: 3125316C8A2A13B8@112.xxx.xx.xxx. CSeq: 55 ACK. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Content-Length: 0. . # U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060 INVITE sip:0123456@my.domain SIP/2.0. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>. Call-ID: 3125316C8A2A13B8@112.xxx.xx.xxx. CSeq: 56 INVITE. Contact: <sip:02xxxxxxxxx@112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>. Proxy-Authorization: Digest username="02xxxxxxxxx", realm="my.domain", nonce="900b46a0-8d88-11de-a6a1-098738f35adb", uri="sip:0123456@my.domain", response="276b44e261c13bd17218adff1150f414", algorithm=MD5, cnonce="CADBE5D624516E8A", qop=auth, nc=00000001. Max-Forwards: 70. Expires: 120. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Supported: 100rel,replaces,timer. Allow-Events: telephone-event,refer. Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH. Content-Type: application/sdp. Accept: application/sdp, multipart/mixed. Accept-Encoding: identity. Content-Length: 359. . v=0. o=user 16423733 16423733 IN IP4 112.xxx.xx.xxx. s=call. c=IN IP4 112.xxx.xx.xxx. t=0 0. m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101. a=sendrecv. a=rtpmap:2 G726-32/8000. a=rtpmap:102 G726-32/8000. a=rtpmap:100 G726-40/8000. a=rtpmap:99 G726-24/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=rtcp:7079. # U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>. Call-ID: 3125316C8A2A13B8@112.xxx.xx.xxx. CSeq: 56 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Content-Length: 0. . # U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>;tag=83t7967K2mXHF. Call-ID: 3125316C8A2A13B8@112.xxx.xx.xxx. CSeq: 56 INVITE. Contact: <sip:0123456@182.xxx.xx.xxx:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 249. . v=0. o=FreeSWITCH 1250746184 1250746185 IN IP4 182.xxx.xx.xxx. s=FreeSWITCH. c=IN IP4 182.xxx.xx.xxx. t=0 0. m=audio 26670 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. # U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060 ACK sip:0123456@182.xxx.xx.xxx:5060;transport=udp SIP/2.0. Via: SIP/2.0/udp 112.xxx.xx.xxx:5060;branch=z9hG4bK39E0B6E237E408FC. From: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. To: <sip:0123456@my.domain>;tag=83t7967K2mXHF. Call-ID: 3125316C8A2A13B8@112.xxx.xx.xxx. CSeq: 56 ACK. Contact: <sip:02xxxxxxxxx@112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>. Max-Forwards: 70. User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009). Content-Length: 0. . # U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060 INVITE sip:02xxxxxxxxx@112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3 SIP/2.0. Via: SIP/2.0/UDP 182.xxx.xx.xxx;rport;branch=z9hG4bK7rpNeaN67y9ta. Max-Forwards: 70. From: <sip:0123456@my.domain>;tag=83t7967K2mXHF. To: <sip:02xxxxxxxxx@my.domain>;tag=9A806878F0882CFC. Call-ID: 3125316C8A2A13B8@112.xxx.xx.xxx. CSeq: 119268091 INVITE. Contact: <sip:0123456@182.xxx.xx.xxx:5060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 249. . v=0. o=FreeSWITCH 1250746184 1250746186 IN IP4 182.xxx.xx.xxx. s=FreeSWITCH. c=IN IP4 182.xxx.xx.xxx. t=0 0. m=audio 26670 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:30.
This is a bug in the fritzbox... you have to set your codec neg. to greedy on the sofia profile and that should fix it. /b On Aug 20, 2009, at 8:35 AM, Peter P GMX wrote:
Hello,
when calling from Fritzbox to a Snom Phone , sound is fine. But when calling an internal Freeswitch number (conference, mailbox) i hear a very choppy voice coming from the fritzbox side. I think it may have to do with the ptime 20msec/30msec.
Example: When calling from the fritzbox to a voicemail then the annoucement from Freeswitch is choppy (too slow with interrups), but the recorded message is fine.
Did anybody experience the same problem?
Best regards Peter
Hello Brian, I have added <param name="inbound-codec-negotiation" value="greedy"/> to internal and external conf. We have only allowed one codec now (G711A) Conferencing MOH is fine. However hearing the other party (Fritzbox is almost impossible). Hearing voicemail announcements is also very choppy with seconds of delay. So, in one case it works fine, in other s not. Any more hints? Best regards Peter Brian West schrieb:
This is a bug in the fritzbox... you have to set your codec neg. to greedy on the sofia profile and that should fix it.
/b
On Aug 20, 2009, at 8:35 AM, Peter P GMX wrote:
Hello,
when calling from Fritzbox to a Snom Phone , sound is fine. But when calling an internal Freeswitch number (conference, mailbox) i hear a very choppy voice coming from the fritzbox side. I think it may have to do with the ptime 20msec/30msec.
Example: When calling from the fritzbox to a voicemail then the annoucement from Freeswitch is choppy (too slow with interrups), but the recorded message is fine.
Did anybody experience the same problem?
Best regards Peter
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Besides taking a hammer to it? Have you tried to make sure you have the latest firmware? Try setting the ptime on the fritz to 20ms? I really can't trust a product that has fritz in its name... it might go on the fritz :P pun intended. /b On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:
Any more hints?
Hello Brian, yes we have updated to the latest Fritzbox Firmware. These FritzBoxes are widely spread here in Germany. I know of a SIP provider who has > 5 Mio FritzBoxes out there. So overall I expect about 10Mio FritzBoxes in Germany, and they are covering a big stake of in the market. So they generally they work. I tested mine against my Asterisk without problems. But in my Freeswitch environment this is not working, and we have manage to couple of these Boxes. So any help is appreciated. Best regards Peter Brian West schrieb:
Besides taking a hammer to it? Have you tried to make sure you have the latest firmware? Try setting the ptime on the fritz to 20ms?
I really can't trust a product that has fritz in its name... it might go on the fritz :P pun intended.
/b
On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:
Any more hints?
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On Thu, Aug 20, 2009 at 1:54 PM, Peter P GMX <Prometheus001@gmx.net> wrote:
Hello Brian,
yes we have updated to the latest Fritzbox Firmware. These FritzBoxes are widely spread here in Germany. I know of a SIP provider who has > 5 Mio FritzBoxes out there. So overall I expect about 10Mio FritzBoxes in Germany, and they are covering a big stake of in the market. So they generally they work. I tested mine against my Asterisk without problems.
But in my Freeswitch environment this is not working, and we have manage to couple of these Boxes. So any help is appreciated.
Just curious - if it seems to be working with Asterisk but not FreeSWITCH then could you do some tcpdumps of working vs. non-working calls and then analyze them with Wireshark? I think Jason Garland's ClueCon presentation(s) might be applicable here. Thoughts? -MC
Best regards Peter
Brian West schrieb:
Besides taking a hammer to it? Have you tried to make sure you have the latest firmware? Try setting the ptime on the fritz to 20ms?
I really can't trust a product that has fritz in its name... it might go on the fritz :P pun intended.
/b
On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:
Any more hints?
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On Thu, 2009-08-20 at 14:35 -0700, Michael Collins wrote:
Just curious - if it seems to be working with Asterisk but not FreeSWITCH then could you do some tcpdumps of working vs. non-working calls and then analyze them with Wireshark? I think Jason Garland's ClueCon presentation(s) might be applicable here.
Just to deepen the mystery a little, we have a FRITZ!Box here in Greece, and it works like a little champ for us. Firmware's 06.04.49, it's talking to a FreeSWITCH box in London. It's set to pick its own codec (but the other end only supports G.711), VAD's off. Cheers -- Dave
Hello Michael, I made some tests with Freeswitch and Fritzbox and found by Wireshark that: within one call * Freeswitch starts sending 20msec packets, then after ~0,2 second sends 30msec packets * FritzBox always sends 30msec packets. The average jitter is below 2 msec in both directions. The below logs shows that Freeswitch considers the FritzBox to be broken and starts using 30msec packets. But there is no SIP message from FS to Fritzbox telling him that FB will use 30msec packets. SDP from FS to Fritzbox always shows ptime:20 BTW: We can ship you a FritzBox if you need one for testing. Best regards Peter Log: 2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms 2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP: v=0 o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx s=FreeSWITCH c=IN IP4 182.xxx.xx.xxx t=0 0 m=audio 30290 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer sofia/internal/02xxxxxxxxx@fs1.my.domain! 2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/02xxxxxxxxx@fs1.my.domain [BREAK] EXECUTE sofia/internal/02xxxxxxxxx@fs1.my.domain playback(voicemail/8000/vm-that_was_an_invalid_ext.wav) 2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16@8000hz 1 channels 20ms 2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel sofia/internal/02xxxxxxxxx@fs1.my.domain entering state [early][183] 2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649 sofia/internal/02xxxxxxxxx@fs1.my.domain receive message [TRANSCODING_NECESSARY] 2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to use ptime 20 but what they meant to say was 30 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen..
Try setting that in your sip profile: <param name="rtp-autofix-timing" value="false" /> Thats a feature to work around with devices lying about their ptime in their sdp payload. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 21-Aug-09, at 11:38 AM, Peter P GMX wrote:
Hello Michael,
I made some tests with Freeswitch and Fritzbox and found by Wireshark that: within one call
* Freeswitch starts sending 20msec packets, then after ~0,2 second sends 30msec packets * FritzBox always sends 30msec packets.
The average jitter is below 2 msec in both directions.
The below logs shows that Freeswitch considers the FritzBox to be broken and starts using 30msec packets. But there is no SIP message from FS to Fritzbox telling him that FB will use 30msec packets. SDP from FS to Fritzbox always shows ptime:20
BTW: We can ship you a FritzBox if you need one for testing.
Best regards Peter
Log:
2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms 2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP: v=0 o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx s=FreeSWITCH c=IN IP4 182.xxx.xx.xxx t=0 0 m=audio 30290 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer sofia/internal/02xxxxxxxxx@fs1.my.domain! 2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/02xxxxxxxxx@fs1.my.domain [BREAK] EXECUTE sofia/internal/02xxxxxxxxx@fs1.my.domain playback(voicemail/8000/vm-that_was_an_invalid_ext.wav) 2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16@8000hz 1 channels 20ms 2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel sofia/internal/02xxxxxxxxx@fs1.my.domain entering state [early][183] 2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649 sofia/internal/02xxxxxxxxx@fs1.my.domain receive message [TRANSCODING_NECESSARY] 2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to use ptime 20 but what they meant to say was 30 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen..
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Hello Mathieu, thank for your help. But this however didn't change the behaviour. I've read of a patch in mod_sofia.c which partly corrects the problem temporarily: When I change Line 784 to if (switch_rtp_ready(tech_pvt->rtp_session) && codec_ms != tech_pvt->codec_ms) { to if (switch_rtp_ready(tech_pvt->rtp_session) && codec_ms != tech_pvt->codec_ms && 0) { (add a "&& 0") to deactivate this expression) the announcements are played correctly to the Fritzbox. Connections to other SIP phones (Snom) are also fine. However the person at the Fritzbox still sounds very choppy in a conference, but this is another module where I do not have a patch available. Best regards Peter Mathieu Rene schrieb:
Try setting that in your sip profile:
<param name="rtp-autofix-timing" value="false" />
Thats a feature to work around with devices lying about their ptime in their sdp payload.
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 21-Aug-09, at 11:38 AM, Peter P GMX wrote:
Hello Michael,
I made some tests with Freeswitch and Fritzbox and found by Wireshark that: within one call
* Freeswitch starts sending 20msec packets, then after ~0,2 second sends 30msec packets * FritzBox always sends 30msec packets.
The average jitter is below 2 msec in both directions.
The below logs shows that Freeswitch considers the FritzBox to be broken and starts using 30msec packets. But there is no SIP message from FS to Fritzbox telling him that FB will use 30msec packets. SDP from FS to Fritzbox always shows ptime:20
BTW: We can ship you a FritzBox if you need one for testing.
Best regards Peter
Log:
2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms 2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP: v=0 o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx s=FreeSWITCH c=IN IP4 182.xxx.xx.xxx t=0 0 m=audio 30290 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer sofia/internal/02xxxxxxxxx@fs1.my.domain! 2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/02xxxxxxxxx@fs1.my.domain [BREAK] EXECUTE sofia/internal/02xxxxxxxxx@fs1.my.domain playback(voicemail/8000/vm-that_was_an_invalid_ext.wav) 2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16@8000hz 1 channels 20ms 2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel sofia/internal/02xxxxxxxxx@fs1.my.domain entering state [early][183] 2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649 sofia/internal/02xxxxxxxxx@fs1.my.domain receive message [TRANSCODING_NECESSARY] 2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to use ptime 20 but what they meant to say was 30 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen..
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try setting FS to 30ms too edit vars.xml and add @30i to everywhere you see PCMU or PCMA so it looks like PCMU@30i from: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> to: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU@30i,PCMA@30i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@30i,PCMA@30i ,GSM"/> On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian@freeswitch.org> wrote:
You can ship me one whois bkw.org, I can add it to my lab.
/b
On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
BTW: We can ship you a FritzBox if you need one for testing.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Hello Anthony, I set PCMA@30i,PCMU@30i and I can see in the logs that PCMA is used. However ptime is set to 20 msec as shown in the Logs: 2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP: v=0 o=user 2075230 2075230 IN IP4 217.xx.xx.xxx s=call c=IN IP4 217.xx.xx.xxx t=0 0 m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101 a=rtpmap:2 G726-32/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:100 G726-40/8000 a=rtpmap:99 G726-24/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7079 2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404 (sofia/internal/02xxxxxxxxx@fs1.my.domain) State NEW 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[G722:9:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec sofia/internal/02xxxxxxxxx@fs1.my.domain PCMA/8000 20 ms 160 samples 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf payload to 101 Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is horrible. Best regards Peter Anthony Minessale schrieb:
try setting FS to 30ms too
edit vars.xml and add @30i to everywhere you see PCMU or PCMA so it looks like PCMU@30i
from:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
to:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU@30i,PCMA@30i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@30i,PCMA@30i,GSM"/>
On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian@freeswitch.org <mailto:brian@freeswitch.org>> wrote:
You can ship me one whois bkw.org <http://bkw.org>, I can add it to my lab.
/b
On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
> > BTW: We can ship you a FritzBox if you need one for testing.
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 ------------------------------------------------------------------------
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That is the remote sdp, not the local sdp. They are sending ptime 20, not us. Are they actually sending 20 ms packets or are they sending 30? MIke On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote:
Hello Anthony,
I set PCMA@30i,PCMU@30i and I can see in the logs that PCMA is used. However ptime is set to 20 msec as shown in the Logs:
2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP: v=0 o=user 2075230 2075230 IN IP4 217.xx.xx.xxx s=call c=IN IP4 217.xx.xx.xxx t=0 0 m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101 a=rtpmap:2 G726-32/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:100 G726-40/8000 a=rtpmap:99 G726-24/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7079
2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404 (sofia/internal/02xxxxxxxxx@fs1.my.domain) State NEW 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[G722:9:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec sofia/internal/02xxxxxxxxx@fs1.my.domain PCMA/8000 20 ms 160 samples 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf payload to 101
Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is horrible.
Best regards Peter
Anthony Minessale schrieb:
try setting FS to 30ms too
edit vars.xml and add @30i to everywhere you see PCMU or PCMA so it looks like PCMU@30i
from:
<X-PRE-PROCESS cmd="set" data ="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
to:
<X-PRE-PROCESS cmd="set" data = "global_codec_prefs =G7221@32000h,G7221@16000h,G722,PCMU@30i,PCMA@30i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@30i,PCMA@30i,GSM"/>
On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian@freeswitch.org <mailto:brian@freeswitch.org>> wrote:
You can ship me one whois bkw.org <http://bkw.org>, I can add it to my lab.
/b
On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
BTW: We can ship you a FritzBox if you need one for testing.
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 ------------------------------------------------------------------------
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mr fritz is lying somewhere get a pcap of the traffic from fritz to FS and look at the size of the audio packets if they are 160(172 with headers) bytes then it's 20ms if it's 240 (252) then it's 30ms if it's saying 20 but it means 30 you should leave the last change in place and also add in <param name="rtp-autofix-timing" value="true" /> and <param name="inbound-codec-negotiation" value="scrooge"/> to overcome the bug on their end. On Mon, Aug 24, 2009 at 2:05 PM, Michael Jerris <mike@jerris.com> wrote:
That is the remote sdp, not the local sdp. They are sending ptime 20, not us. Are they actually sending 20 ms packets or are they sending 30?
MIke
On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote:
Hello Anthony,
I set PCMA@30i,PCMU@30i and I can see in the logs that PCMA is used. However ptime is set to 20 msec as shown in the Logs:
2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP: v=0 o=user 2075230 2075230 IN IP4 217.xx.xx.xxx s=call c=IN IP4 217.xx.xx.xxx t=0 0 m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101 a=rtpmap:2 G726-32/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:100 G726-40/8000 a=rtpmap:99 G726-24/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:7079
2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404 (sofia/internal/02xxxxxxxxx@fs1.my.domain) State NEW 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[G722:9:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec sofia/internal/02xxxxxxxxx@fs1.my.domain PCMA/8000 20 ms 160 samples 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf payload to 101
Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is horrible.
Best regards Peter
Anthony Minessale schrieb:
try setting FS to 30ms too
edit vars.xml and add @30i to everywhere you see PCMU or PCMA so it looks like PCMU@30i
from:
<X-PRE-PROCESS cmd="set" data ="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
to:
<X-PRE-PROCESS cmd="set" data = "global_codec_prefs =G7221@32000h,G7221@16000h,G722,PCMU@30i,PCMA@30i,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@30i,PCMA@30i,GSM"/>
On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian@freeswitch.org <mailto:brian@freeswitch.org>> wrote:
You can ship me one whois bkw.org <http://bkw.org>, I can add it to my lab.
/b
On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
BTW: We can ship you a FritzBox if you need one for testing.
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
pstn:213-799-1400 ------------------------------------------------------------------------
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (7)
-
Anthony Minessale
-
Brian West
-
David Knell
-
Mathieu Rene
-
Michael Collins
-
Michael Jerris
-
Peter P GMX