Shall I know why you want use TCP instead of UDP ? On Tue, Jul 16, 2013 at 9:46 AM, Khoa Pham <onmyway133 at gmail.com> wrote: > Hi, for the SIP transport, pjsip and linphone server do support UDP, TCP > But for Media transport, only UDP is supported. If you 'd like to support > RTP over TCP, please see transport_udp.c in pjmedia project, and make your > own transport_tcp > > Basically, you will not use UDP socket, but TCP socket for media packet > > > On Thu, Apr 11, 2013 at 11:49 AM, Techie Sup <techsup0073 at gmail.com>wrote: > >> I too work on similar feature on routing RTP through TCP and please >> share your observations if you come across with a working solution. >> >> Warm Regards >> Selvam >> >> On 4/7/13, gaurav.srivastava2 at agnity.com <gaurav.srivastava2 at agnity.com> >> wrote: >> > >> > Please check http://www.pjsip.org/sip_media_features.htm [1] >> > >> > On Sun, 7 >> > Apr 2013 01:05:49 -0700, Haomiao Huang >> > wrote: >> >> Sorry, what I meant was >> > that the RTP packets were still going out over UDP >> >> and not TCP. So the >> > SIP messaging packets were definitely TCP, but the >> >> media packets were >> > still UDP. >> >> >> >> Any thoughts on why this might be the case? Do I need to do >> > something >> >> separate in PJSIP to make the RTP go out over TCP as well? >> >> >> >> >> > >> >> On Sat, Apr 6, 2013 at 6:08 AM, J Mena wrote: >> >> >> >>> RTP is the protocol >> > used by the application, the rtp packets can be sent >> >>> over different >> > transpors, udp or tcp >> >>> >> >>> >> >>> >> >>> On Apr 5, 2013, at 10:24 PM, "Haomiao >> > Huang" >> >>> wrote: >> >>> >> >>> Hi, >> >>> I'm trying to get a VOIP app working over >> > 3G. It looks like the RTP >> >>> packets going to and from the phone are not >> > getting through. I would like >> >>> to send media using TCP. How would I go >> > about doing this, and does the >> >>> linphone server support this? I'm using >> > sip.linphone.org for my >> >>> registration. >> >>> >> >>> I've enabled TCP transport >> > on both clients (standard linphone app on iOS >> >>> and pjsua on MacOS). The >> > SIP packets go through on TCP but RTP is still >> >>> being used for the the >> > audio itself. >> >>> >> >>> Any ideas/thoughts on this? Is it the server that's >> > causing this, or do >> >>> I need to do something on the client side? Is this >> > even a good idea? Any >> >>> help would be appreciated! >> >>> -Haomiao >> >>> >> >>> >> >>> >> > _______________________________________________ >> >>> Visit our blog: >> > http://blog.pjsip.org >> >>> >> >>> pjsip mailing list >> >>> pjsip at lists.pjsip.org >> >>> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >>> >> >>> >> >>> >> > _______________________________________________ >> >>> Visit our blog: >> > http://blog.pjsip.org >> >>> >> >>> pjsip mailing list >> >>> pjsip at lists.pjsip.org >> >>> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >>> >> >>> >> > >> > >> > Links: >> > ------ >> > [1] >> > http://www.pjsip.org/sip_media_features.htm >> > >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> > > > > -- > Khoa Pham > HCMC University of Science > www.fantageek.com > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -- Regards, J Alex Antony Vijay. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20130718/cfdc8b67/attachment-0001.html>
