RTCP in ICE candidate list

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Sure,you can check  PJMEDIA_ADVERTISE_RTCP  in your config file,by default
,it is set to true,
and have a look at this line  if (PJMEDIA_ADVERTISE_RTCP &&
!acc_cfg->ice_cfg.ice_no_rtcp)

hope this will help.

Best Regard,

Frank.zheng


2013/12/13 Dmytro Bogovych <dmytro.bogovych at gmail.com>

> Greetings.
>
> I test the own implementation of softphone. It is not based on pjsip;
> instead it uses combination of resiprocate/jrtplib/own ice & media stacks.
>
> I try to make ice traversal working for rtcp component too (rtp component
> works good).
>
> The test peer is pjsua from 2.1.0 on Ubuntu 12.04 in virtualbox.
>
> The sent offer is:
> <---INVITE sip:dbogovych1 at voipobjects.com SIP/2.0
> <---Via: SIP/2.0/TCP 192.168.1.102:5060
> ;branch=z9hG4bK-524287-1---a27da3758f4a7f37;rport
> <---Max-Forwards: 70
> <---Contact: <sip:dbogovych at 95.132.162.61
> :5060;transport=tcp>;+sip.instance="8078730"
> <---To: <sip:dbogovych1 at voipobjects.com>
> <---From: <sip:dbogovych@xxxxxxxxxxxxxxxxxxxxx>;tag=ee352f5f
> <---Call-ID: KcLpGZhqZzfz033okcroYg..
> <---CSeq: 1 INVITE
> <---Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, INFO, MESSAGE,
> REFER, NOTIFY, SUBSCRIBE, REGISTER
> <---Content-Type: application/sdp
> <---Supported: timer, norefersub, replaces, eventlist
> <---User-Agent: IntTalk_2.2.10
> <---Content-Length: 991
> <---
> <---v=0
> <---o=ITS_user 0 1 IN IP4 8046
> <---s=ITS_session
> <---c=IN IP4 95.132.162.61
> <---t=0 0
> <---a=ice-pwd:sxknaaiewaqajgizkrkfaz
> <---a=ice-ufrag:itab
> <---m=audio 8046 RTP/AVP 106 0 8 3 100 99 9 97 103 104 101
> <---a=rtpmap:106 opus/16000
> <---a=rtpmap:0 pcmu/8000
> <---a=rtpmap:8 pcma/8000
> <---a=rtpmap:3 gsm/8000
> <---a=rtpmap:100 ilbc/8000
> <---a=rtpmap:99 ilbc/8000
> <---a=fmtp:99 mode=20
> <---a=rtpmap:9 g722/16000
> <---a=rtpmap:97 isac/16000
> <---a=rtpmap:103 speex/8000
> <---a=rtpmap:104 speex/16000
> <---a=rtpmap:101 telephone-event/8000
> <---a=silenceSupp:off - - - -
> <---a=RS:0
> <---a=RR:0
> <---a=candidate:20490432 1 UDP 2113929471 192.168.56.1 8046 typ host
> <---a=candidate:1711384768 1 UDP 2113929471 192.168.1.102 8046 typ host
> <---a=candidate:1728161984 1 UDP 1677721855 95.132.162.61 8046 typ srflx
> raddr 192.168.1.102 rport 8046
> <---a=candidate:20490432 2 UDP 2113929470 192.168.56.1 8047 typ host
> <---a=candidate:1711384768 2 UDP 2113929470 192.168.1.102 8047 typ host
> <---a=candidate:1728161984 2 UDP 1677721854 95.132.162.61 8047 typ srflx
> raddr 192.168.1.102 rport 8047
>
>
> The answer is:
> --->SIP/2.0 200 OK
> --->Via: SIP/2.0/TCP 192.168.1.102:5060
> ;rport=2712;received=95.132.162.61;branch=z9hG4bK-524287-1---a27da3758f4a7f37
> --->Record-Route: <sip:0.0.0.0;lr;r2=on>
> --->Record-Route: <sip:0.0.0.0;transport=tcp;lr;r2=on>
> --->Contact: <sip:dbogovych1 at 95.132.162.61:59679;ob>;+sip.ice
> --->To: <sip:dbogovych1 at voipobjects.com
> >;tag=JVWGJPZQrdfCP4-Xf0.t41FCQP1lbC1X
> --->From: <sip:dbogovych@xxxxxxxxxxxxxxxxxxxxx>;tag=ee352f5f
> --->Call-ID: KcLpGZhqZzfz033okcroYg..
> --->CSeq: 1 INVITE
> --->Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
> NOTIFY, REFER, MESSAGE, OPTIONS
> --->Content-Type: application/sdp
> --->Supported: replaces, 100rel, timer, norefersub
> --->Content-Length: 461
> --->
> --->v=0
> --->o=- 3595921123 3595921124 IN IP4 95.132.162.61
> --->s=pjmedia
> --->b=AS:84
> --->t=0 0
> --->a=X-nat:8
> --->m=audio 50889 RTP/AVP 0 101
> --->c=IN IP4 95.132.162.61
> --->b=TIAS:64000
> --->b=RS:0
> --->b=RR:0
> --->a=sendrecv
> --->a=rtpmap:0 PCMU/8000
> --->a=ice-ufrag:761e30d2
> --->a=ice-pwd:350aa3bd
> --->a=candidate:Sa00020f 1 UDP 1862270975 95.132.162.61 50889 typ srflx
> raddr 10.0.2.15 rport 4037
> --->a=candidate:Ha00020f 1 UDP 1694498815 10.0.2.15 4037 typ host
> --->a=rtpmap:101 telephone-event/8000
> --->a=fmtp:101 0-15
>
> Why pjsua does not insert RTCP component information into candidate list?
> Does it support RTCP in ICE?
>
> Thank you!
>
>
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