[pjsip] RTCP in ICE candidate list

Dmytro Bogovych dmytro.bogovych at gmail.com
Fri Dec 13 06:09:52 EST 2013


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 at license.crypttalk.com>;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 at license.crypttalk.com>;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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