[Sip-implementors] requirement to play media stream while on hold?
If you are placed on hold (a=sendonly), and the holder sends you a media stream, are you required to play it? I looked through RFC's 3261, 2327, and 3264, but I don't really see a requirement to actually play a media stream ever- when on hold or not. -- Paul Heitkemper Software Engineer [AI] 9701 Taylorsville Rd. Louisville, KY 40299 office: 1-502-267-7436 x1674 direct: 1-502-287-0069 e-mail: paul.heitkem...@atlasied.com<mailto:paul.heitkem...@iedaudio.com> www.AtlasIED.com ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] RTP with wrong payload
RFC 3550 Section 5.1 " A receiver MUST ignore packets with payload types that it does not understand." -- Paul On 2018-11-14 01:30, Sundbaum Per-Johan (Telenor Sverige AB) wrote: > Thanks ! > Ok, but to be more specific, are there any general recommendations on > question like: Should the PBX drop the packets with wrong payload and let the > call/session continue or should the PBX also drop the entire call ? > BR/pj > > -Original Message- > From: Dale R. Worley [mailto:wor...@ariadne.com] > Sent: den 14 november 2018 05:25 > To: Sundbaum Per-Johan (Telenor Sverige AB) > Cc: sip-implementors@lists.cs.columbia.edu > Subject: Re: [Sip-implementors] RTP with wrong payload > > "Sundbaum Per-Johan (Telenor Sverige AB)" > writes: > >> Can someone help me getting a link to information regarding how a PBX should >> handle the case when for example the codec G.711A is agreed upon, but >> somehow there are a couple of G.722 packets received that have the same >> ports on UDP level, but clearly belongs to another RTP stream when you look >> at sequence numbers, timestamps and SSRC ? > > There's no real definition of "clearly belongs to". If a packet arrives at a > port, it is processed according to the usage that has been negotiated for > that port. > > Dale > ___ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors [1] Links: -- [1] https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
[Sip-implementors] sip trunking question
I hope this is apropos for this listserv. Is there a good source somewhere which describes (with examples) the messaging inherent in making a trunk? What does it mean to be a trunk? Thanks, Paul ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
[Sip-implementors] ***SPAM*** trunk information needed
I'm having trouble understanding how a trunk operates at the messaging level. Can someone please point me to some information that could help me? Thanks, Paul ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
[Sip-implementors] send audio to multicast address?
Can I send SDP to a phone to request it to send audio to a multicast Group, as opposed to the UAC in the Contact URL? Does anyone have any idea of how widely-implemented this is in the industry? Thanks, Paul ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
