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Problem starting evaluation
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28.03.2008 16:23 |
raphael.david@online.auditec-newtest.com |
Hello,
I am testeting TERASIP in order to include it in our product.
Unfortunately it does not work for a simple sip acount.
Here is a part of the trace file showing that INVITE command failed:
**************************************************************************
>>>>INVITE sip:0153246669@freephonie.net SIP/2.0
Via: SIP/2.0/UDP 193.253.88.88:52060;branch=z9hG4bK-d8754z-b43d186f3f384922-1---d8754z-;rport
Max-Forwards: 70
Contact:
To:
From: "0952725555";tag=5c56cc46
Call-ID: Y2FlYmMzYTRmNTk5YTM4ZDNkNDdlOGQxNmYxNzcyMzE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, INVITE, INFO, OPTIONS, NOTIFY, OPTIONS, NOTIFY
Content-Type: application/sdp
User-Agent: TERASIP/1.0.0.941
Content-Length: 382
v=0
o=- 1206718775 1206718775 IN IP4 193.253.88.88
s=TERASIP
c=IN IP4 193.253.88.88
t=0 0
m=audio 52236 RTP/AVP 0 8 3 113 114 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:113 iLBC/8000
a=fmtp:113 mode=20
a=rtpmap:114 iLBC/8000
a=fmtp:114 mode=30
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<<<< SIP/2.0 423 Interval Too Brief
Via: SIP/2.0/UDP 193.253.88.88:52060;received=193.253.88.88;rport=51476;branch=z9hG4bK-d8754z-1673045dd20f3312-1---d8754z-
To: "0952725555" ;tag=00-32583-009a5c3d-2febe67c2
From: "0952725555" ;tag=8a75f041
Call-ID: MzliMjA1Mjc3ZTM4MTFkZTFlOWJiMzcwMDkxYWRjOGY.
CSeq: 1 REGISTER
Min-Expires: 1800
Content-Length: 0
***************************************************************************
(I used the SamplePhone VB application)
Can you help me ?
Do you need the complete trace files ?
An other question : do you support G729 and G722 codec ?
Thanks,
Best regards,
--------------------------
Raphael DAVID
AUDITEC S.A. www.auditec-newtest.com
raphael.david@auditec-newtest.com
--------------------------
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28.03.2008 21:30 |
TERASENS Support |
Hi Raphael,
the SIP response does not match the INVITE message. In fact it is a response
to
a REGISTER request. I think the SIP server complains because the
registration expiration
value is too low. We will provide a new version with a configurable
expiration value
next week, so please hang on.
As to codec support. G.729 support requires licensing the codec from
siprolabs.com.
G.722 support might be added more easily, please contact me via private mail
on this
issue.
--
Best regards,
Matthias Moetje
-------------------------------------
TERASENS GmbH
Augustenstraße 24
80333 Munich, GERMANY
-------------------------------------
e-mail: moetje at terasens dot com
www: www.terasens.com
-------------------------------------
wrote in message
news:950725e7-34e2-448e-9c9b-9acdf5ccd429@news.terasens.de...
>
> Hello,
>
> I am testeting TERASIP in order to include it in our product.
>
> Unfortunately it does not work for a simple sip acount.
>
> Here is a part of the trace file showing that INVITE command failed:
>
> **************************************************************************
>>>>>INVITE sip:0153246669@freephonie.net SIP/2.0
> Via: SIP/2.0/UDP
> 193.253.88.88:52060;branch=z9hG4bK-d8754z-b43d186f3f384922-1---d8754z-;rport
> Max-Forwards: 70
> Contact:
> To:
> From: "0952725555";tag=5c56cc46
> Call-ID: Y2FlYmMzYTRmNTk5YTM4ZDNkNDdlOGQxNmYxNzcyMzE.
> CSeq: 1 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, INVITE, INFO, OPTIONS,
> NOTIFY, OPTIONS, NOTIFY
> Content-Type: application/sdp
> User-Agent: TERASIP/1.0.0.941
> Content-Length: 382
>
> v=0
> o=- 1206718775 1206718775 IN IP4 193.253.88.88
> s=TERASIP
> c=IN IP4 193.253.88.88
> t=0 0
> m=audio 52236 RTP/AVP 0 8 3 113 114 110 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:113 iLBC/8000
> a=fmtp:113 mode=20
> a=rtpmap:114 iLBC/8000
> a=fmtp:114 mode=30
> a=rtpmap:110 speex/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=sendrecv
>
>
> <<<< SIP/2.0 423 Interval Too Brief
> Via: SIP/2.0/UDP
> 193.253.88.88:52060;received=193.253.88.88;rport=51476;branch=z9hG4bK-d8754z-1673045dd20f3312-1---d8754z-
> To: "0952725555"
> ;tag=00-32583-009a5c3d-2febe67c2
> From: "0952725555" ;tag=8a75f041
> Call-ID: MzliMjA1Mjc3ZTM4MTFkZTFlOWJiMzcwMDkxYWRjOGY.
> CSeq: 1 REGISTER
> Min-Expires: 1800
> Content-Length: 0
>
>
> ***************************************************************************
>
> (I used the SamplePhone VB application)
>
> Can you help me ?
> Do you need the complete trace files ?
>
>
> An other question : do you support G729 and G722 codec ?
>
> Thanks,
>
> Best regards,
>
>
> --------------------------
> Raphael DAVID
> AUDITEC S.A. www.auditec-newtest.com
> raphael.david@auditec-newtest.com
> --------------------------
>
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08.04.2008 10:12 |
raphael.david@online.auditec-newtest.com |
Hi Matthias,
Thank you for your answer.
Ok I will wait for next version to test freephonie provider.
Now I am testing connection with an Asterisk server (TrixBox).
Connection works, but I do not get any audio data, IMemoryData buffer size is always 0.
(The test program works fine with an other TAPI provider connected to the same SIP server)
Is it a limitation of the free licence ?
Best regards,
--------------------------
Raphael DAVID
AUDITEC S.A.
raphael.david@auditec-newtest.com
--------------------------
>Hi Raphael,
>
>the SIP response does not match the INVITE message. In fact it is a response
>to
>a REGISTER request. I think the SIP server complains because the
>registration expiration
>value is too low. We will provide a new version with a configurable
>expiration value
>next week, so please hang on.
>
>As to codec support. G.729 support requires licensing the codec from
>siprolabs.com.
>G.722 support might be added more easily, please contact me via private mail
>on this
>issue.
>
>
>--
>Best regards,
>
>Matthias Moetje
>-------------------------------------
>TERASENS GmbH
>Augustenstraße 24
>80333 Munich, GERMANY
>-------------------------------------
>e-mail: moetje at terasens dot com
>www: www.terasens.com
>-------------------------------------
>
> wrote in message
>news:950725e7-34e2-448e-9c9b-9acdf5ccd429@news.terasens.de...
>>
>> Hello,
>>
>> I am testeting TERASIP in order to include it in our product.
>>
>> Unfortunately it does not work for a simple sip acount.
>>
>> Here is a part of the trace file showing that INVITE command failed:
>>
>> **************************************************************************
>>>>>>INVITE sip:0153246669@freephonie.net SIP/2.0
>> Via: SIP/2.0/UDP
>> 193.253.88.88:52060;branch=z9hG4bK-d8754z-b43d186f3f384922-1---d8754z-;rport
>> Max-Forwards: 70
>> Contact:
>> To:
>> From: "0952725555";tag=5c56cc46
>> Call-ID: Y2FlYmMzYTRmNTk5YTM4ZDNkNDdlOGQxNmYxNzcyMzE.
>> CSeq: 1 INVITE
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, INVITE, INFO, OPTIONS,
>> NOTIFY, OPTIONS, NOTIFY
>> Content-Type: application/sdp
>> User-Agent: TERASIP/1.0.0.941
>> Content-Length: 382
>>
>> v=0
>> o=- 1206718775 1206718775 IN IP4 193.253.88.88
>> s=TERASIP
>> c=IN IP4 193.253.88.88
>> t=0 0
>> m=audio 52236 RTP/AVP 0 8 3 113 114 110 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:113 iLBC/8000
>> a=fmtp:113 mode=20
>> a=rtpmap:114 iLBC/8000
>> a=fmtp:114 mode=30
>> a=rtpmap:110 speex/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=sendrecv
>>
>>
>> <<<< SIP/2.0 423 Interval Too Brief
>> Via: SIP/2.0/UDP
>> 193.253.88.88:52060;received=193.253.88.88;rport=51476;branch=z9hG4bK-d8754z-1673045dd20f3312-1---d8754z-
>> To: "0952725555"
>> ;tag=00-32583-009a5c3d-2febe67c2
>> From: "0952725555" ;tag=8a75f041
>> Call-ID: MzliMjA1Mjc3ZTM4MTFkZTFlOWJiMzcwMDkxYWRjOGY.
>> CSeq: 1 REGISTER
>> Min-Expires: 1800
>> Content-Length: 0
>>
>>
>> ***************************************************************************
>>
>> (I used the SamplePhone VB application)
>>
>> Can you help me ?
>> Do you need the complete trace files ?
>>
>>
>> An other question : do you support G729 and G722 codec ?
>>
>> Thanks,
>>
>> Best regards,
>>
>>
>> --------------------------
>> Raphael DAVID
>> AUDITEC S.A. www.auditec-newtest.com
>> raphael.david@auditec-newtest.com
>> --------------------------
>>
>
> |
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14.04.2008 05:00 |
TERASENS Support |
Raphael,
sorry for the late reply, our server had a crash during the last few days.
Could you please post a code snippet of how you are creating the samples and
doing buffer allocation?
We have this kind of stream data access working in our applications, so I am
sure we will be able to get this fixed.
Best regards,
Matthias Moetje
-------------------------------------
TERASENS GmbH
Augustenstraße 24
80333 Munich, GERMANY
-------------------------------------
e-mail: moetje at terasens dot com
www: www.terasens.com
-------------------------------------
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