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[Prometheus001 at gmx.net]

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.






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[brian at freeswitch.org]

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:

[Prometheus001 at gmx.net]

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:

[brian at freeswitch.org]

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 pun intended.

/b

On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:

[Prometheus001 at gmx.net]

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:

[msc at freeswitch.org]

On Thu, Aug 20, 2009 at 1:54 PM, Peter P GMX <Prometheus001@gmx.net (Prometheus001@gmx.net)> wrote:

[dave at 3c.co.uk]

On Thu, 2009-08-20 at 14:35 -0700, Michael Collins wrote:

[Prometheus001 at gmx.net]

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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[mrene_lists at avgs.ca]

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:

[Prometheus001 at gmx.net]

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:

[brian at freeswitch.org]

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:

[anthony.minessale at g...]

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 (brian@freeswitch.org)> wrote:

[Prometheus001 at gmx.net]

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:

[mike at jerris.com]

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:

[anthony.minessale at g...]

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 (mike@jerris.com)> wrote: