[pjsip] RTCP in ICE candidate list

Dmytro Bogovych dmytro.bogovych at gmail.com
Mon Dec 16 11:13:21 EST 2013


I have another question related to the ice/rtp/sdp area.

Does pjsip support rtp/rtcp multiplexing? rtcp-mux attribute?
Thank you :)


On Mon, Dec 16, 2013 at 1:44 PM, Dmytro Bogovych
<dmytro.bogovych at gmail.com>wrote:

> Thank you, your advice really helped.
>
> Yes, it is defined to 1.
> Therefore RTCP candidate should be in candidate list.
> However pjsua alters this behavior; the account/ice settings prevent on
> this.
>
> I think there is issue. The --help shows me --ice-no-rtcp is not default
> on. But actually it is turned on - i checked via dc command.
>
>
>
> On Sat, Dec 14, 2013 at 8:36 AM, Yuming Zheng <
> zhengyumingnanjing at gmail.com> wrote:
>
>> 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 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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>>>
>>>
>>
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>>
>
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