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:sip_qu...@yahoo.co.in] Sent: Monday, April 27, 2009 9:06 PM To: Shamik Saha (WT01 - Telecom Equipment) Cc: sip-implement...@cs.columbia.edu 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, shamik.s...@wipro.com <shamik.s...@wipro.com> wrote: From: shamik.s...@wipro.com <shamik.s...@wipro.com> Subject: RE: [Sip-implementors] Query for SDP Negotiation To: sip_qu...@yahoo.co.in Cc: sip-implement...@cs.columbia.edu 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:sip_qu...@yahoo.co.in] Sent: Monday, April 27, 2009 8:46 PM To: Shamik Saha (WT01 - Telecom Equipment) Cc: sip-implementors-boun...@lists.cs.columbia.edu 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, shamik.s...@wipro.com <shamik.s...@wipro.com> wrote: From: shamik.s...@wipro.com <shamik.s...@wipro.com> Subject: RE: [Sip-implementors] Query for SDP Negotiation To: sip_qu...@yahoo.co.in Cc: sip-implementors-boun...@lists.cs.columbia.edu 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: sip-implementors-boun...@lists.cs.columbia.edu <http://in.mc88.mail.yahoo.com/mc/compose?to=sip-implementors-boun...@lists.cs.columbia.edu> [mailto:sip-implementors-boun...@lists.cs.columbia.edu <http://in.mc88.mail.yahoo.com/mc/compose?to=sip-implementors-boun...@lists.cs.columbia.edu> ] 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) <somesh.shanb...@nsn.com <http://in.mc88.mail.yahoo.com/mc/compose?to=somesh.shanb...@nsn.com> > wrote: From: Shanbhag, Somesh (NSN - IN/Bangalore) <somesh.shanb...@nsn.com <http://in.mc88.mail.yahoo.com/mc/compose?to=somesh.shanb...@nsn.com> > Subject: Re: [Sip-implementors] Query for SDP Negotiation To: "ext Peter Nijhuis" <peter.nijh...@televersal.com <http://in.mc88.mail.yahoo.com/mc/compose?to=peter.nijh...@televersal.com> >, "ext friend friend" <sip_qu...@yahoo.co.in <http://in.mc88.mail.yahoo.com/mc/compose?to=sip_qu...@yahoo.co.in> >, "sip fourm" <sip-implementors@lists.cs.columbia.edu <http://in.mc88.mail.yahoo.com/mc/compose?to=sip-implement...@lists.cs.columbia.edu> > 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:peter.nijh...@televersal.com <http://in.mc88.mail.yahoo.com/mc/compose?to=peter.nijh...@televersal.com> ] 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: sip-implementors-boun...@lists.cs.columbia.edu <http://in.mc88.mail.yahoo.com/mc/compose?to=sip-implementors-boun...@lists.cs.columbia.edu> [mailto:sip- > implementors-boun...@lists.cs.columbia.edu <http://in.mc88.mail.yahoo.com/mc/compose?to=implementors-boun...@lists.cs.columbia.edu> ] 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: sip-implementors-boun...@lists.cs.columbia.edu <http://in.mc88.mail.yahoo.com/mc/compose?to=sip-implementors-boun...@lists.cs.columbia.edu> [mailto:sip- > implementors-boun...@lists.cs.columbia.edu <http://in.mc88.mail.yahoo.com/mc/compose?to=implementors-boun...@lists.cs.columbia.edu> ] 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. 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