It is the UAS who is creating the non-standard answer, in which case, there is no way the UAC can send 4xx response. Also, I do see a common codec which are iLBC and telephone-event. The callee should have used the same dynamic payloads for these codecs as in offer, but it chose difference ones. In my opinion, the caller should not terminate the dialog as the RFC says SHOULD and not MUST. RFC 3264 In the case of RTP, if a particular codec was referenced with a specific payload type number in the offer, that same payload type number SHOULD be used for that codec in the answer. Even if the same payload type number is used, the answer MUST contain rtpmap attributes to define the payload type mappings for dynamic payload types, and SHOULD contain mappings for static payload types. The media formats in the "m=" line MUST be listed in order of preference, with the first format listed being preferred. In this case, preferred means that the offerer SHOULD use the format with the highest preference from the answer.
On Mon, Apr 27, 2009 at 7:26 AM, friend friend <[email protected]> wrote: > 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]> wrote: > > > From: Shanbhag, Somesh (NSN - IN/Bangalore) <[email protected]> > Subject: Re: [Sip-implementors] Query for SDP Negotiation > To: "ext Peter Nijhuis" <[email protected]>, "ext friend friend" > <[email protected]>, "sip fourm" <[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]] > 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] [mailto:sip- >> [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] [mailto:sip- >> [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/ >> _______________________________________________ >> Sip-implementors mailing list >> [email protected] >> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors >> >> _______________________________________________ >> Sip-implementors mailing list >> [email protected] >> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > Explore your hobbies and interests. Go to > http://in.promos.yahoo.com/groups/ > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
