Hi Vijay,
you should check whether the same rtpmap value is being used by both the
offerer and the answerer for the dynamic payloads,for the dynamic payloads the
value should be in the range 96-127,and atleast one codec should overlap among
the negotiated codec.
the payload type mappings from rfc 1890 are as follows,you can try to match it
with your scenario,since you are facing this problem only with dynamic payload
types,so perhaps this will help you
PT encoding audio/video clock rate channels
name (A/V) (Hz) (audio)
_______________________________________________________________
0 PCMU A 8000 1
1 1016 A 8000 1
2 G721 A 8000 1
3 GSM A 8000 1
4 unassigned A 8000 1
5 DVI4 A 8000 1
6 DVI4 A 16000 1
7 LPC A 8000 1
8 PCMA A 8000 1
9 G722 A 8000 1
10 L16 A 44100 2
11 L16 A 44100 1
12 unassigned A
13 unassigned A
14 MPA A 90000
15 G728 A 8000 1
16--23 unassigned A
24 unassigned V
25 CelB V 90000
26 JPEG V 90000
27 unassigned V
28 nv V 90000
29 unassigned V
30 unassigned V
31 H261 V 90000
32 MPV V 90000
33 MP2T AV 90000
34--71 unassigned ?
72--76 reserved N/A N/A N/A
77--95 unassigned ?
96--127 dynamic ?
Thanks and regards,
Shamik Saha
Thanks and regards,
Shamik Saha
Project Engineer
Voice Protocols
Cell : +91-9886704155
________________________________
From: friend friend [mailto:[email protected]]
Sent: Monday, April 27, 2009 9:06 PM
To: Shamik Saha (WT01 - Telecom Equipment)
Cc: [email protected]
Subject: RE: [Sip-implementors] Query for SDP Negotiation
Could you please tell me, is this the codec problem. bcoz particular codec is
not working
--- On Mon, 27/4/09, [email protected] <[email protected]> wrote:
From: [email protected] <[email protected]>
Subject: RE: [Sip-implementors] Query for SDP Negotiation
To: [email protected]
Cc: [email protected]
Date: Monday, 27 April, 2009, 8:56 PM
No Vijay that is not a problem because the default value is sendrecv so
if you do not specify anything it will be taken as send-recv,so if you want it
to be send-recv it is not mandatory in either offer or answer.
Thanks and regards,
Shamik Saha
Project Engineer
Voice Protocols
Cell : +91-9886704155
________________________________
From: friend friend [mailto:[email protected]]
Sent: Monday, April 27, 2009 8:46 PM
To: Shamik Saha (WT01 - Telecom Equipment)
Cc: [email protected]
Subject: RE: [Sip-implementors] Query for SDP Negotiation
Hi Shamik,
Media negotiation is happening properly. i think the ports are opening
properly bcoz its working for static codecs. but its not working for dynamic
codecs.
In caller side Direction line is not added. but callee is sending sendrecv in
200 OK.
Caller side Direction line is not present. is this the cause of problem?
I think in offer Direction line is not mandatory rite?
Thanks & regards,
vijay
--- On Mon, 27/4/09, [email protected] <[email protected]> wrote:
From: [email protected] <[email protected]>
Subject: RE: [Sip-implementors] Query for SDP Negotiation
To: [email protected]
Cc: [email protected]
Date: Monday, 27 April, 2009, 8:11 PM
Hi Vijay,
There can be many reasons for a one way speech-path,
You need to check the SDP negotiation properly,whether both the sides
are sening send-recv or not,
Are the codecs matching?
Then you need to check your application,whether it is opening the send
and receive ports properly or not.
A look into the negotiated SDP will tell whether the problem is with
the SDP negotiation or the application.
Thanks and regards,
Shamik Saha
Project Engineer
Voice Protocols
Cell : +91-9886704155
-----Original Message-----
From: [email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]>
[mailto:[email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]>
] On Behalf Of friend friend
Sent: Monday, April 27, 2009 7:57 PM
To: ext Peter Nijhuis; sip fourm; Somesh (NSN - IN/Bangalore)Shanbhag
Subject: Re: [Sip-implementors] Query for SDP Negotiation
Dear Folks,
Thank you for useful responses.
Please clarify one more query.
In case, Both side(caller & callee) call is established and RTP(Both
side iLBC) also flowing successfully but one side audio is coming and the other
side audio is not coming.
so what would be the problem(Other than mic bcoz mic is working
properly)?
Thanks & Regards,
vijay
--- On Mon, 27/4/09, Shanbhag, Somesh (NSN - IN/Bangalore)
<[email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]> > wrote:
From: Shanbhag, Somesh (NSN - IN/Bangalore) <[email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]> >
Subject: Re: [Sip-implementors] Query for SDP Negotiation
To: "ext Peter Nijhuis" <[email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]> >,
"ext friend friend" <[email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]> >, "sip
fourm" <[email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]>
>
Date: Monday, 27 April, 2009, 7:09 PM
Yeah, that is even better .. .what I thought was since CALLEE had
already replied not-in-protocol-way, CALLER may terminate the dialog.
Somesh
-----Original Message-----
From: ext Peter Nijhuis [mailto:[email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]> ]
Sent: Monday, April 27, 2009 7:00 PM
To: Shanbhag, Somesh (NSN - IN/Bangalore); ext friend friend; sip fourm
Subject: RE: [Sip-implementors] Query for SDP Negotiation
I think Callee should respond with 415 unsupported media type, since it
is not supporting media types 102, 0, 8 or 106.
Met vriendelijke groet, with kind regards, mit freundlichen Gruß
Peter Nijhuis
> -----Original Message-----
> From: [email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]>
[mailto:sip-
> [email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]>
] On Behalf Of Shanbhag,
> Somesh (NSN - IN/Bangalore)
> Sent: maandag 27 april 2009 15:07
> To: ext friend friend; sip fourm
> Subject: Re: [Sip-implementors] Query for SDP Negotiation
>
> The basic thing is CALLEE has to take the subset of codecs offered by
> CALLER and reply back.
> But in your case, CALLEE is replying with different set of codecs (97
> 101) in reply to CALLER codecs ( 102 0 8 106 ) IMHO, since the
> capabilities mis-match, immdiately end the call using BYE / CANCEL
> which ever is relevant.
>
>
> Somesh
>
> -----Original Message-----
> From: [email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]>
[mailto:sip-
> [email protected]
<http://in.mc88.mail.yahoo.com/mc/[email protected]>
] On Behalf Of ext friend
> friend
> Sent: Monday, April 27, 2009 2:59 PM
> To: sip fourm
> Subject: [Sip-implementors] Query for SDP Negotiation
>
> Dear Folks,
> I have doubt in the following scenario.
>
> Caller's sdp :
>
> v=0
> o=- 1234 1 IN IP4 10.10.20.35
> s=-
> c=IN IP4 10.10.20.35
> t=0 0
> m=audio 12000 RTP/AVP 102 0 8 106
> a=rtpmap:102 iLBC/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 telephone-event/8000
>
>
> Callee's negotiated sdp :
>
> v=0
> o=- 3449811996 3449811996 IN IP4 10.10.20.4 s=SJphone c=IN IP4
> 10.10.20.4 t=0 0 m=audio 49164 RTP/AVP 97 101 c=IN IP4 10.10.20.4
> a=rtpmap:97 iLBC/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=setup:active
> a=sendrecv
>
> In this case,Is callee's negotiation method is wrong?
>
> Callee should send like m=audio 49164 RTP/AVP 102 106 rite?
>
> In this case, after call establishment,
> from caller sending RTP using 102 (UnKnown)
> from callee sending RTP using 97 (iLBC)
>
> So caller hearing callee's audio but callee not able to hear caller's
> audio.
>
> please clarify this issue.
>
> Thanks & Regards,
> vijay
>
>
> Bring your gang together. Do your thing. Find your favourite
>Yahoo! group at http://in.promos.yahoo.com/groups/
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