I think we must allways end the discussion  , it will help another
people in future.

Thanks!

Nahum,
 
 
it a=rtpmap:101 PCMA/8000   is wrong  , you must set to  AVP=8  -->
a=rtpmap:8 PCMA/8000
 

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
89.139.126.116:10313;rport;branch=z9hG4bkoeyj2axhv4z3c4zanbrhl5t5808egte
js
Max-Forwards: 70
From:
sip:[email protected];tag=pn4ix1tv9mmwnj1miezr1v7m6ic
bi7
To: sip:[email protected]
Call-ID: jk0h84pgx73s9fwmib9fum4dg7d67wp7
CSeq: 1001 INVITE
Contact: <sip:[email protected]:10313;transport=UDP>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Authorization: Digest username="0001001103826", realm="88.215.60.14",
nonce="f2b771a85a0c8d1db118a1131ba9e2444b69c2bc",
uri="sip:[email protected]",
response="c25c60f0674dbf00c90ed323ccfacbb5", algorithm=md5
Content-Type: application/sdp
Content-Length: 178
 
v=0
o=- 1265222732 1265222732 IN IP4 89.139.126.116
s=nsip server version 1.0
c=IN IP4 89.139.126.116
t=0 0
m=audio 10313 RTP/AVP 8
a=sendrecv
a=rtpmap:101 PCMA/8000  
 
 
 
 
+++
 
see a offer with several codecs 
 
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.139.9.148:5060;branch=z9hG4bK397447f47c22a401b
Max-Forwards: 70
From: +17867101049
<sip:[email protected]>;tag=49a4aa8d92;epid=SC237417
To: <sip:[email protected];transport=udp>
Call-ID: bd0ba0d842524bf3
CSeq: 1 INVITE
Allow:  INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,UPDATE
Allow-Events: talk, hold, conference
Contact: +17867101049 <sip:[email protected]:5060;transport=udp>
Min-SE: 90
Session-Expires: 300
Supported: timer, replaces, 100rel
User-Agent: optiPoint 420 Standard/V7 V7 R2.3.0
X-Siemens-Call-Type: ST-insecure
Content-Type: application/sdp
Content-Length: 368
 
v=0
o=MxSIP 0 203513255 IN IP4 10.139.9.148
s=SIP Call
c=IN IP4 10.139.9.148
t=0 0
m=audio 5004 RTP/AVP 0 8 9 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:10

 
 
Regards!
Valdemar
 
 

 

-----Original Message-----
From: ext [email protected] [mailto:[email protected]]

Sent: Wednesday, February 03, 2010 11:13 PM
To: Pavesi, Valdemar (NSN - US/Boca Raton); [email protected];
[email protected]
Cc: [email protected]
Subject: RE: [Sip-implementors] What is wrong with the following
INVITE???

Hi,

I think this is the catch. The rtpmap defines the encoding used for
codec "8", but mistakenly you have used "101".

Regards
Sunil Verma

-----Original Message-----
From: [email protected]
[mailto:[email protected]] On Behalf Of
Pavesi, Valdemar (NSN - US/Boca Raton)
Sent: Thursday, February 04, 2010 12:39 AM
To: ext Pranab Bohra; Nahum Nir
Cc: [email protected]
Subject: Re: [Sip-implementors] What is wrong with the following
INVITE???

Nahum,
 
 
it a=rtpmap:101 PCMA/8000   is wrong  , you must set to  AVP=8  -->
a=rtpmap:8 PCMA/8000
 

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
89.139.126.116:10313;rport;branch=z9hG4bkoeyj2axhv4z3c4zanbrhl5t5808egte
js
Max-Forwards: 70
From:
sip:[email protected];tag=pn4ix1tv9mmwnj1miezr1v7m6ic
bi7
To: sip:[email protected]
Call-ID: jk0h84pgx73s9fwmib9fum4dg7d67wp7
CSeq: 1001 INVITE
Contact: <sip:[email protected]:10313;transport=UDP>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Authorization: Digest username="0001001103826", realm="88.215.60.14",
nonce="f2b771a85a0c8d1db118a1131ba9e2444b69c2bc",
uri="sip:[email protected]",
response="c25c60f0674dbf00c90ed323ccfacbb5", algorithm=md5
Content-Type: application/sdp
Content-Length: 178
 
v=0
o=- 1265222732 1265222732 IN IP4 89.139.126.116 s=nsip server version
1.0 c=IN IP4 89.139.126.116 t=0 0 m=audio 10313 RTP/AVP 8 a=sendrecv
a=rtpmap:101 PCMA/8000  
 
 
 
 
+++
 
see a offer with several codecs and codecs listed on sdp param-m
corresponde to list of param-a :

 
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.139.9.148:5060;branch=z9hG4bK397447f47c22a401b
Max-Forwards: 70
From: +17867101049
<sip:[email protected]>;tag=49a4aa8d92;epid=SC237417
To: <sip:[email protected];transport=udp>
Call-ID: bd0ba0d842524bf3
CSeq: 1 INVITE
Allow:  INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,UPDATE
Allow-Events: talk, hold, conference
Contact: +17867101049 <sip:[email protected]:5060;transport=udp>
Min-SE: 90
Session-Expires: 300
Supported: timer, replaces, 100rel
User-Agent: optiPoint 420 Standard/V7 V7 R2.3.0
X-Siemens-Call-Type: ST-insecure
Content-Type: application/sdp
Content-Length: 368
 
v=0
o=MxSIP 0 203513255 IN IP4 10.139.9.148
s=SIP Call
c=IN IP4 10.139.9.148
t=0 0
m=audio 5004 RTP/AVP 0 8 9 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:10

 
 
Regards!
Valdemar 

-----Original Message-----
From: [email protected]
[mailto:[email protected]] On Behalf Of ext
Pavesi, Valdemar (NSN - US/Boca Raton)
Sent: Wednesday, February 03, 2010 11:21 AM
To: ext Pranab Bohra; Nahum Nir
Cc: [email protected]
Subject: Re: [Sip-implementors] What is wrong with the following
INVITE???

Hi,

But it should not be the problem , the endpoint-b  don't nothing about
the same port for sip/rtp.

You must see a RTP-stream coming to your device. (endpoint-a)


Regards!
Valdemar

 

-----Original Message-----
From: [email protected]
[mailto:[email protected]] On Behalf Of ext
Pranab Bohra
Sent: Wednesday, February 03, 2010 11:04 AM
To: Nahum Nir
Cc: [email protected]
Subject: Re: [Sip-implementors] What is wrong with the following
INVITE???

Hi,

You are using port 44074 for both SIP signalling and RTP, generally
media and signalling are handled on different ports.

Cheers,
Pranab

On Wed, Feb 3, 2010 at 3:15 PM, Nahum Nir <[email protected]>
wrote:
> Hi All,
>
> I'm trying to make an INVITE so that the incoming audio will be
received on
> 89.139.126.116:44074.
>
> I get Session In Progress and OK but no incoming audio.
> What is wrong???
>
> Thanks,
> Nahum
>
> INVITE
sip:[email protected]<sip%3a%[email protected].
theiptele.com>SIP/2.0
> Via: SIP/2.0/UDP 89.139.126.116:44074
> ;rport;branch=z9hG4bkxu9h74udx9n9jhm4fp35vfurr7ao7x8nu
> Max-Forwards:
> 70
>
> From:
sip:[email protected]<sip%[email protected]
eiptele.com>;tag=74tosdtd1fjmj03nxkdqjicczu7tw0
>
> To:
sip:[email protected]<sip%3a%[email protected].
theiptele.com>
>
> Call-ID:
> bcu20hjb9solhzl5q70ymf22hxhg10xm
>
> CSeq: 1001
> INVITE
>
> Contact: <sip:[email protected]:44074;transport=UDP>
>
> Authorization: Digest username="0001001103826", realm="88.215.60.14", 
> nonce="8191cd28e8d275daaf2b6b441ac804534b69414e", uri="
> sip:[email protected] <sip%3a%[email protected]>",
> response="63fa61f9eae18f1e0c4bd1be37c30406",
> algorithm=md5
>
> Content-Type:
> application/sdp
>
> Content-Length:
> 176
>
>
> v=0
> o=- 1265189596 1265189596 IN IP4 89.139.126.116 s=nsip server version 
> 1.0 c=IN IP4 89.139.126.116 t=0 0 m=audio 44074 RTP/AVP 8 a=sendrecv
> a=rtpmap:8 PCMA/8000
> _______________________________________________
> Sip-implementors mailing list
> [email protected]
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>
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