[jitsi-users] Call starts cutting out after sending DTMF in call with keypad


#1

Hi All,

I've just started using the latest build of Jitsi x86 on Windows 7.
Connected via SIP to Anveo.

I like it quite a bit, but I've noticed an issue with sending DTMF
within a call. Essentially, as soon as I click a digit on the keypad,
the tone is detected on the other end, but the call audio starts
rapidly cutting out in a rhythmic staccato fashion, to the extent that
it's impossible to comprehend the incoming stream.

DTMF is set to auto. I tried switching it to RTP but then the tones
don't seem to be detected (though RFC 2833 is supported by Anveo). SIP
Input does not work at all. Inband gives the cutting out I mentioned.

Any suggestions? Is this a bug?

Yours,

Dylan

Dylan Gordon
PhD Candidate
Department of Anthropology, University of Toronto
http://twitter.com/@KnowWildFood


#2

Hey Dylan,

Hi All,

I've just started using the latest build of Jitsi x86 on Windows 7.
Connected via SIP to Anveo.

I like it quite a bit, but I've noticed an issue with sending DTMF
within a call. Essentially, as soon as I click a digit on the keypad,
the tone is detected on the other end, but the call audio starts
rapidly cutting out in a rhythmic staccato fashion, to the extent that
it's impossible to comprehend the incoming stream.

Is this incoming or outgoing audio? Can you check if you have the same issue when using ippi.com?

DTMF is set to auto. I tried switching it to RTP but then the tones
don't seem to be detected (though RFC 2833 is supported by Anveo).

Do they advertise support for telephone-event in the SDP offers/answers

SIP
Input does not work at all. Inband gives the cutting out I mentioned.

Any suggestions? Is this a bug?

Could be. Or it could an issue with them. That's why I suggested you try with ippi.com. This should helps us decide.

Emil

···

On 23.11.13, 02:18, Dylan Gordon wrote:

Yours,

Dylan

Dylan Gordon
PhD Candidate
Department of Anthropology, University of Toronto
http://twitter.com/@KnowWildFood

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#3

Thanks for a prompt response.

I call pretty much only SIP to PSTN, by the way. So, I tested the
outgoing and incoming audio calling my cellular. On the Jitsi end, it
sounds like staccato cutting out. On the cellular, I hear the DTMF
tone repeated over and over again with staccato breaks.

Calling SIP:904@mouselike.org via ippi.com works fine.

I can't see anything regarding telephone-event in SDP support for
Anveo. I'm afraid that's a bit beyond my scope of knowledge... But I
would expect if it is relatively standard they support it, they are a
major business VoIP provider/reseller in North America.

So, it looks like perhaps it is Anveo (or something specific to SIP to
PSTN, but I can't call PSTN from ippi without a paid account).
However, their service works fine with every other softphone I have
used (CSipSimple, SIPDroid, Zoiper, Anveo Communicator) so I'm not
sure what is different in the case of Jitsi...

Let me know if you have any other suggestions or things to test. Or of
course I can try another softphone. But I do quite like Jitsi in all
other respects, so thanks for that.

Yours,

Dylan

Dylan Gordon
PhD Candidate
Department of Anthropology, University of Toronto
http://twitter.com/@KnowWildFood

···

On Sat, Nov 23, 2013 at 5:50 AM, Emil Ivov <emcho@jitsi.org> wrote:

Hey Dylan,

On 23.11.13, 02:18, Dylan Gordon wrote:

Hi All,

I've just started using the latest build of Jitsi x86 on Windows 7.
Connected via SIP to Anveo.

I like it quite a bit, but I've noticed an issue with sending DTMF
within a call. Essentially, as soon as I click a digit on the keypad,
the tone is detected on the other end, but the call audio starts
rapidly cutting out in a rhythmic staccato fashion, to the extent that
it's impossible to comprehend the incoming stream.

Is this incoming or outgoing audio? Can you check if you have the same issue
when using ippi.com?

DTMF is set to auto. I tried switching it to RTP but then the tones
don't seem to be detected (though RFC 2833 is supported by Anveo).

Do they advertise support for telephone-event in the SDP offers/answers

SIP
Input does not work at all. Inband gives the cutting out I mentioned.

Any suggestions? Is this a bug?

Could be. Or it could an issue with them. That's why I suggested you try
with ippi.com. This should helps us decide.

Emil

Yours,

Dylan

Dylan Gordon
PhD Candidate
Department of Anthropology, University of Toronto
http://twitter.com/@KnowWildFood

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

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#4

Well just tried with a test account there and things worked for me with both RTP and inband DTMF. Didn't notice any issues :frowning:

···

On 23.11.13, 14:48, Dylan Gordon wrote:

Thanks for a prompt response.

I call pretty much only SIP to PSTN, by the way. So, I tested the
outgoing and incoming audio calling my cellular. On the Jitsi end, it
sounds like staccato cutting out. On the cellular, I hear the DTMF
tone repeated over and over again with staccato breaks.

Calling SIP:904@mouselike.org via ippi.com works fine.

I can't see anything regarding telephone-event in SDP support for
Anveo. I'm afraid that's a bit beyond my scope of knowledge... But I
would expect if it is relatively standard they support it, they are a
major business VoIP provider/reseller in North America.

So, it looks like perhaps it is Anveo (or something specific to SIP to
PSTN, but I can't call PSTN from ippi without a paid account).
However, their service works fine with every other softphone I have
used (CSipSimple, SIPDroid, Zoiper, Anveo Communicator) so I'm not
sure what is different in the case of Jitsi...

Let me know if you have any other suggestions or things to test. Or of
course I can try another softphone. But I do quite like Jitsi in all
other respects, so thanks for that.

Yours,

Dylan

Dylan Gordon
PhD Candidate
Department of Anthropology, University of Toronto
http://twitter.com/@KnowWildFood

On Sat, Nov 23, 2013 at 5:50 AM, Emil Ivov <emcho@jitsi.org> wrote:

Hey Dylan,

On 23.11.13, 02:18, Dylan Gordon wrote:

Hi All,

I've just started using the latest build of Jitsi x86 on Windows 7.
Connected via SIP to Anveo.

I like it quite a bit, but I've noticed an issue with sending DTMF
within a call. Essentially, as soon as I click a digit on the keypad,
the tone is detected on the other end, but the call audio starts
rapidly cutting out in a rhythmic staccato fashion, to the extent that
it's impossible to comprehend the incoming stream.

Is this incoming or outgoing audio? Can you check if you have the same issue
when using ippi.com?

DTMF is set to auto. I tried switching it to RTP but then the tones
don't seem to be detected (though RFC 2833 is supported by Anveo).

Do they advertise support for telephone-event in the SDP offers/answers

SIP
Input does not work at all. Inband gives the cutting out I mentioned.

Any suggestions? Is this a bug?

Could be. Or it could an issue with them. That's why I suggested you try
with ippi.com. This should helps us decide.

Emil

Yours,

Dylan

Dylan Gordon
PhD Candidate
Department of Anthropology, University of Toronto
http://twitter.com/@KnowWildFood

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

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#5

Hmmm, strange. Well, I guess the most pragmatic thing to do is switch
to something else.... Perhaps it's the proxy (I use sip.ca.anveo.com
vs. sip.anveo.com), maybe it's my PC, 32 bit vs 64 bit. Lots of
variables I don't have time to test. Though do let me know if there's
anything else you'd like to know.

Thanks for your help!

Yours,

Dylan

Dylan Gordon
PhD Candidate
Department of Anthropology, University of Toronto
http://twitter.com/@KnowWildFood

···

On Sun, Nov 24, 2013 at 3:53 AM, Emil Ivov <emcho@jitsi.org> wrote:

Well just tried with a test account there and things worked for me with both
RTP and inband DTMF. Didn't notice any issues :frowning:

On 23.11.13, 14:48, Dylan Gordon wrote:

Thanks for a prompt response.

I call pretty much only SIP to PSTN, by the way. So, I tested the
outgoing and incoming audio calling my cellular. On the Jitsi end, it
sounds like staccato cutting out. On the cellular, I hear the DTMF
tone repeated over and over again with staccato breaks.

Calling SIP:904@mouselike.org via ippi.com works fine.

I can't see anything regarding telephone-event in SDP support for
Anveo. I'm afraid that's a bit beyond my scope of knowledge... But I
would expect if it is relatively standard they support it, they are a
major business VoIP provider/reseller in North America.

So, it looks like perhaps it is Anveo (or something specific to SIP to
PSTN, but I can't call PSTN from ippi without a paid account).
However, their service works fine with every other softphone I have
used (CSipSimple, SIPDroid, Zoiper, Anveo Communicator) so I'm not
sure what is different in the case of Jitsi...

Let me know if you have any other suggestions or things to test. Or of
course I can try another softphone. But I do quite like Jitsi in all
other respects, so thanks for that.

Yours,

Dylan

Dylan Gordon
PhD Candidate
Department of Anthropology, University of Toronto
http://twitter.com/@KnowWildFood

On Sat, Nov 23, 2013 at 5:50 AM, Emil Ivov <emcho@jitsi.org> wrote:

Hey Dylan,

On 23.11.13, 02:18, Dylan Gordon wrote:

Hi All,

I've just started using the latest build of Jitsi x86 on Windows 7.
Connected via SIP to Anveo.

I like it quite a bit, but I've noticed an issue with sending DTMF
within a call. Essentially, as soon as I click a digit on the keypad,
the tone is detected on the other end, but the call audio starts
rapidly cutting out in a rhythmic staccato fashion, to the extent that
it's impossible to comprehend the incoming stream.

Is this incoming or outgoing audio? Can you check if you have the same
issue
when using ippi.com?

DTMF is set to auto. I tried switching it to RTP but then the tones
don't seem to be detected (though RFC 2833 is supported by Anveo).

Do they advertise support for telephone-event in the SDP offers/answers

SIP
Input does not work at all. Inband gives the cutting out I mentioned.

Any suggestions? Is this a bug?

Could be. Or it could an issue with them. That's why I suggested you try
with ippi.com. This should helps us decide.

Emil

Yours,

Dylan

Dylan Gordon
PhD Candidate
Department of Anthropology, University of Toronto
http://twitter.com/@KnowWildFood

_______________________________________________
users mailing list
users@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/users

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#6

Hi Dylan,

I have an Anveo account and I'll try dialing an Ivr with options to see
what happens with jitsi.

···

--------------
sip:jungleboogie@sip2sip.info
inum: +883510009902611
On Nov 24, 2013 6:02 AM, "Dylan Gordon" <dylan.gordon@gmail.com> wrote:

Hmmm, strange. Well, I guess the most pragmatic thing to do is switch
to something else.... Perhaps it's the proxy (I use sip.ca.anveo.com
vs. sip.anveo.com), maybe it's my PC, 32 bit vs 64 bit. Lots of
variables I don't have time to test. Though do let me know if there's
anything else you'd like to know.

Thanks for your help!

Yours,

Dylan

Dylan Gordon
PhD Candidate
Department of Anthropology, University of Toronto
http://twitter.com/@KnowWildFood

On Sun, Nov 24, 2013 at 3:53 AM, Emil Ivov <emcho@jitsi.org> wrote:
> Well just tried with a test account there and things worked for me with
both
> RTP and inband DTMF. Didn't notice any issues :frowning:
>
>
> On 23.11.13, 14:48, Dylan Gordon wrote:
>>
>> Thanks for a prompt response.
>>
>> I call pretty much only SIP to PSTN, by the way. So, I tested the
>> outgoing and incoming audio calling my cellular. On the Jitsi end, it
>> sounds like staccato cutting out. On the cellular, I hear the DTMF
>> tone repeated over and over again with staccato breaks.
>>
>> Calling SIP:904@mouselike.org via ippi.com works fine.
>>
>> I can't see anything regarding telephone-event in SDP support for
>> Anveo. I'm afraid that's a bit beyond my scope of knowledge... But I
>> would expect if it is relatively standard they support it, they are a
>> major business VoIP provider/reseller in North America.
>>
>> So, it looks like perhaps it is Anveo (or something specific to SIP to
>> PSTN, but I can't call PSTN from ippi without a paid account).
>> However, their service works fine with every other softphone I have
>> used (CSipSimple, SIPDroid, Zoiper, Anveo Communicator) so I'm not
>> sure what is different in the case of Jitsi...
>>
>> Let me know if you have any other suggestions or things to test. Or of
>> course I can try another softphone. But I do quite like Jitsi in all
>> other respects, so thanks for that.
>>
>> Yours,
>>
>> Dylan
>>
>>
>> Dylan Gordon
>> PhD Candidate
>> Department of Anthropology, University of Toronto
>> http://twitter.com/@KnowWildFood
>>
>>
>> On Sat, Nov 23, 2013 at 5:50 AM, Emil Ivov <emcho@jitsi.org> wrote:
>>>
>>> Hey Dylan,
>>>
>>>
>>> On 23.11.13, 02:18, Dylan Gordon wrote:
>>>>
>>>>
>>>> Hi All,
>>>>
>>>> I've just started using the latest build of Jitsi x86 on Windows 7.
>>>> Connected via SIP to Anveo.
>>>>
>>>> I like it quite a bit, but I've noticed an issue with sending DTMF
>>>> within a call. Essentially, as soon as I click a digit on the keypad,
>>>> the tone is detected on the other end, but the call audio starts
>>>> rapidly cutting out in a rhythmic staccato fashion, to the extent that
>>>> it's impossible to comprehend the incoming stream.
>>>
>>>
>>>
>>> Is this incoming or outgoing audio? Can you check if you have the same
>>> issue
>>> when using ippi.com?
>>>
>>>
>>>> DTMF is set to auto. I tried switching it to RTP but then the tones
>>>> don't seem to be detected (though RFC 2833 is supported by Anveo).
>>>
>>>
>>>
>>> Do they advertise support for telephone-event in the SDP offers/answers
>>>
>>>
>>>> SIP
>>>> Input does not work at all. Inband gives the cutting out I mentioned.
>>>>
>>>> Any suggestions? Is this a bug?
>>>
>>>
>>>
>>> Could be. Or it could an issue with them. That's why I suggested you
try
>>> with ippi.com. This should helps us decide.
>>>
>>> Emil
>>>>
>>>>
>>>>
>>>> Yours,
>>>>
>>>> Dylan
>>>>
>>>>
>>>> Dylan Gordon
>>>> PhD Candidate
>>>> Department of Anthropology, University of Toronto
>>>> http://twitter.com/@KnowWildFood
>>>>
>>>> _______________________________________________
>>>> users mailing list
>>>> users@jitsi.org
>>>> Unsubscribe instructions and other list options:
>>>> http://lists.jitsi.org/mailman/listinfo/users
>>>>
>>>
>>> --
>>> https://jitsi.org
>>
>>
>
> --
> https://jitsi.org

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