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MatzeMuc86 | 1 Mar 2011 08:40
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Re: Reference Implementation of a SIP Server

OK, but who is doing the conference (including Mixing) in PJSIP? The Client?

If I understand it the right way: I need a Sip registrar Server and in
addition an special (PJSIP) client which is able to do the conference
stuff). At this situation there is no registration server needed anymore
(until the next call starts)?

Thanks
MatzeMuc86

-----Ursprüngliche Nachricht-----
Von: Klaus Darilion [mailto:klaus.mailinglists <at> pernau.at] 
Gesendet: Montag, 28. Februar 2011 19:25
An: pjsip list
Cc: MatzeMuc86
Betreff: Re: [pjsip] Reference Implementation of a SIP Server

Am 28.02.2011 16:08, schrieb MatzeMuc86:
> Hi,
> 
>  
> 
> I get more used to the PJSIP project and begin to like it - especially 
> the multi-abstraction layer library!!!
> 
> I'm still not an expert with PJSIP and the possibilities but could not 
> find a running SIP Server. Is there such a server for which I am too 
> blind to find or is there not such a SIP server but it is possible to
build it, e..
> with the PJSUA library?
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Ming | 1 Mar 2011 08:50
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Re: Compiling PJSIP 1.8.5 for use on iOS 3.x and 4.x

Hi Even,

Thanks for the suggestion. This is a good idea indeed. In fact, we add
the multitasking support detection in the pjsua-lib instead so that
the application does not need to worry about this anymore.

For more details, please refer to ticket #1203
(http://trac.pjsip.org/repos/ticket/1203)

Regards,
Ming

On Wed, Feb 23, 2011 at 8:21 PM, Even André Fiskvik <eaf <at> oyatel.com> wrote:
> Thanks for the pointer, we just needed to use this one ourselves to make our work with an iPhone 3G.
> Should this perhaps be added to the ipjsua reference code?
> The device capability test could be checked with [UIDevice multitaskingSupported].
>
> Best regards,
> Even André
>
> On 9. nov. 2010, at 18.13, Ming wrote:
>
>> Hi Brad,
>>
>> For devices that do not support multitasking, you have to disable the
>> bg feature using: pj_activesock_enable_iphone_os_bg(PJ_FALSE)
>>
>> Regards,
>> Ming
>>
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Saúl Ibarra Corretgé | 1 Mar 2011 09:06
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Re: Reference Implementation of a SIP Server

On 02/28/2011 04:08 PM, MatzeMuc86 wrote:
> Hi,
>
> I get more used to the PJSIP project and begin to like it – especially
> the multi-abstraction layer library!!!
>
> I’m still not an expert with PJSIP and the possibilities but could not
> find a running SIP Server. Is there such a server for which I am too
> blind to find or is there not such a SIP server but it is possible to
> build it, e.. with the PJSUA library?
>

Its not clear to me what you are trying to accomplish. Do you need a SIP 
server to test a client you want to build? Then Klaus gave you the answer.

If you want to create a SIP server with PJSIP, you'll find a sample 
stateful and stateless proxy inside the PJSIP source code.

Regards,

--

-- 
Saúl Ibarra Corretgé
AG Projects
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s.marek | 1 Mar 2011 09:56
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RFC 3326

Hello everyone,

the other day I was having a look at the sources in the pjsip/src/pjsip 
folder in order to evaluate the Reason: header on an incoming CANCEL. The 
RFC behind the idea is RFC 3326. To make things clear: we're not talking 
about the reason string within the status line. RFC 3326 talks about a 
seperate header line named Reason.

Apart from Reason not having a specific handler in sip_parser.c it seems 
to me, that up to inv_respond_incoming_cancel() in sip_inv.c I have access 
to the received data through rdata with the Reason header somewhere inside 
rdata->msg_info. But my on_call_state() handler is called later, after 
PJSIP has responded to the CANCEL message. As far as I can see, I don't 
have access to the received headers at that point anymore.

I don't have enough insight to provide a patch, so forgive me this shot in 
the dark: pjsip_endpt_create_response() in sip_util.c does copy some 
things over from rdata->msg_info to tdata. That might be a starting point 
to save the received Reason header into the tx_data struct and start to 
support RFC 3326? That way my on_call_state() callback might have access 
to the Reason header through the body of the event struct given to it?

Or does PJSIP support RFC 3326 already and I was just too ignorant to find 
the right spots? I'm working with a pretty current (2 weeks old) version 
of the 1.8.10 branch.

Any other thoughts on that matter?

Thanks in advance,
Sebastian.
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matthias@infinatic.de | 1 Mar 2011 13:22
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SDP negotiation done, message body is ignored

Hello,
i`ve developed a application with pjsua and tested them with a local sip-server and everything works fine but if i try to make a call with a extern sip server only my conversational partner can hear me, and no rtp is sended by him. In debug messages i can see:

Making call with acc #0 to sip:0030343467824 <at> 194.138.116.14
Media index 0 selected for call 2
pjsua_core.c  TX 1101 bytes Request msg INVITE/cseq=12614 (tdta0xc81ba00) to UDP 194.138.116.14:5060:
Via: SIP/2.0/UDP 79.197.201.235:45242;rport;branch=z9hG4bKPjmzNISAbsYUjezdOw.BKUNkjrNo.O4ILm
Max-Forwards: 70
Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV
CSeq: 12614 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: SIP-Phone v0.1/darwin
Content-Type: application/sdp
Content-Length:   426

v=0
o=- 3507969578 3507969578 IN IP4 10.13.45.237
s=pjmedia
c=IN IP4 10.13.45.237
t=0 0
a=X-nat:0
m=audio 8004 RTP/AVP 98 97 99 104 3 0 8 96
a=rtcp:8005 IN IP4 10.13.45.237
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15

--end msg--

os_core_unix.c  Info: possibly re-registering existing thread
12:59:38.868   pjsua_core.c  RX 301 bytes Response msg 100/INVITE/cseq=12614 (rdata0xd80e214) from UDP 194.138.116.14:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 79.197.201.235:45242;branch=z9hG4bKPjmzNISAbsYUjezdOw.BKUNkjrNo.O4ILm;rport=45242
Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV
CSeq: 12614 INVITE


--end msg--
12:59:41.265   pjsua_core.c  RX 831 bytes Response msg 180/INVITE/cseq=12614 (rdata0x6812814) from UDP 194.138.116.14:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 79.197.201.235:45242;branch=z9hG4bKPjmzNISAbsYUjezdOw.BKUNkjrNo.O4ILm;rport=45242
To: <sip:0030343467824 <at> 194.138.116.14>;tag=SEC11-a7aa8c0-1e7aa8c0-1-1aJ2ZxBVt57P
Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV
CSeq: 12614 INVITE
X-Siemens-Call-Type: ST-insecure
Date: Tue, 01 Mar 2011 11:59:41 GMT
Content-Type: application/sdp
Content-Length: 298

v=0
o=- 1298983958 1298983958 IN IP4 194.138.116.14
s=RG8700 0lx0486 0x00500101
c=IN IP4 194.138.116.14
t=0 0
a=sendrecv
a=pmft:T38
m=audio 30172 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sqn:0
a=cdsc:1 image udptl t38
a=ptime:20
a=sendrecv

--end msg--
12:59:41.266        pjsua.c  Call 2 state changed to EARLY (180 Ringing)
12:59:41.267  strm0x607c774  VAD temporarily disabled
12:59:41.267  strm0x607c774  Encoder stream started
12:59:41.267  strm0x607c774  Decoder stream started
12:59:41.268  pjsua_media.c  Media updates, stream #0: PCMA (sendrecv)
12:59:41.268   conference.c  Port 3 (sip:0030343467824 <at> 194.138.116.14) transmitting to port 0 (iPhone IO device)
12:59:41.268   conference.c  Port 0 (iPhone IO device) transmitting to port 3 (sip:0030343467824 <at> 194.138.116.14)
12:59:41.269        pjsua.c  Media for call 2 is active
12:59:41.895  strm0x607c774  VAD re-enabled
12:59:43.374   pjsua_core.c  RX 1043 bytes Response msg 200/INVITE/cseq=12614 (rdata0x608bc14) from UDP 194.138.116.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 79.197.201.235:45242;branch=z9hG4bKPjmzNISAbsYUjezdOw.BKUNkjrNo.O4ILm;rport=45242
To: <sip:0030343467824 <at> 194.138.116.14>;tag=SEC11-a7aa8c0-1e7aa8c0-1-1aJ2ZxBVt57P
Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV
CSeq: 12614 INVITE
X-Siemens-Call-Type: ST-insecure
Accept-Language: en;q=0.0
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
Require: timer
Session-Expires: 1800;refresher=uas
Supported: timer
Date: Tue, 01 Mar 2011 11:59:43 GMT
Content-Type: application/sdp
Content-Length: 298

v=0
o=- 1298983958 1298983958 IN IP4 194.138.116.14
s=RG8700 0lx0486 0x00500101
c=IN IP4 194.138.116.14
t=0 0
a=sendrecv
a=pmft:T38
m=audio 30172 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sqn:0
a=cdsc:1 image udptl t38
a=ptime:20
a=sendrecv

12:59:43.375        pjsua.c  Call 2 state changed to CONNECTING
12:59:43.375   inv0xc819864  SDP negotiation done, message body is ignored
12:59:43.376   pjsua_core.c  TX 417 bytes Request msg ACK/cseq=12614 (tdta0x608f000) to UDP 194.138.116.14:5060:
Via: SIP/2.0/UDP 79.197.201.235:45242;rport;branch=z9hG4bKPjkB9AY5itqdhldn-V0u5WAIFw-QKfoDsx
Max-Forwards: 70
Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV
CSeq: 12614 ACK
Content-Length:  0

12:59:43.376        pjsua.c  Call 2 state changed to CONFIRMED
  
12:59:47.167   pjsua_core.c  TX 452 bytes Request msg BYE/cseq=12615 (tdta0x6092000) to UDP 194.138.116.14:5060:
Via: SIP/2.0/UDP 79.197.201.235:45242;rport;branch=z9hG4bKPjdO8FSLaxJeuskHWBsewdOUMLnLGhrSqy
Max-Forwards: 70
Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV
CSeq: 12615 BYE
User-Agent: SIP-Phone v0.1/darwin
Content-Length:  0

I think the line "SDP negotiation done, message body is ignored" should tell me whats wrong but i cant see the problem.. 
can anybody say me whats the problem?
im grateful for every help i can get 
best regards Matthias 
<div>Hello,<div>i`ve developed a application with pjsua and tested them with a local sip-server and everything works fine but if i try to make a call with a extern sip server only my conversational partner&nbsp;can hear me, and no rtp is sended by him. In debug messages i can see:</div>
<div><span class="Apple-style-span"><span class="Apple-style-span"><br></span></span></div>
<div><div>
<div>Making call with acc #0 to <a href="sip:0030343467824 <at> 194.138.116.14">sip:0030343467824 <at> 194.138.116.14</a>
</div>
<div>Media index 0 selected for call 2</div>
<div>pjsua_core.c&nbsp; TX 1101 bytes Request msg INVITE/cseq=12614 (tdta0xc81ba00) to UDP 194.138.116.14:5060:</div>
<div>INVITE <a href="sip:0030343467824 <at> 194.138.116.14">sip:0030343467824 <at> 194.138.116.14</a> SIP/2.0</div>
<div>Via: SIP/2.0/UDP 79.197.201.235:45242;rport;branch=z9hG4bKPjmzNISAbsYUjezdOw.BKUNkjrNo.O4ILm</div>
<div>Max-Forwards: 70</div>
<div>From: <a href="sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7">sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7</a>
</div>
<div>To: <a href="sip:0030343467824 <at> 194.138.116.14">sip:0030343467824 <at> 194.138.116.14</a>
</div>
<div>Contact: &lt;<a href="sip:49894423416532 <at> 79.197.201.235:45242;transport=UDP;ob">sip:49894423416532 <at> 79.197.201.235:45242;transport=UDP;ob</a>&gt;</div>
<div>Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV</div>
<div>CSeq: 12614 INVITE</div>
<div>Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS</div>
<div>Supported: replaces, 100rel, timer, norefersub</div>
<div>Session-Expires: 1800</div>
<div>Min-SE: 90</div>
<div>User-Agent: SIP-Phone v0.1/darwin</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: &nbsp; 426</div>
<div><br></div>
<div>v=0</div>
<div>o=- 3507969578 3507969578 IN IP4 10.13.45.237</div>
<div>s=pjmedia</div>
<div>c=IN IP4 10.13.45.237</div>
<div>t=0 0</div>
<div>a=X-nat:0</div>
<div>m=audio 8004 RTP/AVP 98 97 99 104 3 0 8 96</div>
<div>a=rtcp:8005 IN IP4 10.13.45.237</div>
<div>a=rtpmap:98 speex/16000</div>
<div>a=rtpmap:97 speex/8000</div>
<div>a=rtpmap:99 speex/32000</div>
<div>a=rtpmap:104 iLBC/8000</div>
<div>a=fmtp:104 mode=30</div>
<div>a=rtpmap:3 GSM/8000</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:8 PCMA/8000</div>
<div>a=sendrecv</div>
<div>a=rtpmap:96 telephone-event/8000</div>
<div>a=fmtp:96 0-15</div>
<div><br></div>
<div>--end msg--</div>
<div><br></div>
<div>os_core_unix.c&nbsp; Info: possibly re-registering existing thread</div>
<div>12:59:38.868 &nbsp; pjsua_core.c&nbsp; RX 301 bytes Response msg 100/INVITE/cseq=12614 (rdata0xd80e214) from UDP 194.138.116.14:5060:</div>
<div>SIP/2.0 100 Trying</div>
<div>Via: SIP/2.0/UDP 79.197.201.235:45242;branch=z9hG4bKPjmzNISAbsYUjezdOw.BKUNkjrNo.O4ILm;rport=45242</div>
<div>From: <a href="sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7">sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7</a>
</div>
<div>To: <a href="sip:0030343467824 <at> 194.138.116.14">sip:0030343467824 <at> 194.138.116.14</a>
</div>
<div>Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV</div>
<div>CSeq: 12614 INVITE</div>
<div><br></div>
<div><br></div>
<div>--end msg--</div>
<div>12:59:41.265 &nbsp; pjsua_core.c&nbsp; RX 831 bytes Response msg 180/INVITE/cseq=12614 (rdata0x6812814) from UDP 194.138.116.14:5060:</div>
<div>SIP/2.0 180 Ringing</div>
<div>Via: SIP/2.0/UDP 79.197.201.235:45242;branch=z9hG4bKPjmzNISAbsYUjezdOw.BKUNkjrNo.O4ILm;rport=45242</div>
<div>From: <a href="sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7">sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7</a>
</div>
<div>To: &lt;<a href="sip:0030343467824 <at> 194.138.116.14">sip:0030343467824 <at> 194.138.116.14</a>&gt;;tag=SEC11-a7aa8c0-1e7aa8c0-1-1aJ2ZxBVt57P</div>
<div>Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV</div>
<div>CSeq: 12614 INVITE</div>
<div>Contact: &lt;<a href="sip:0030343467824 <at> 194.138.116.14:5060;transport=udp">sip:0030343467824 <at> 194.138.116.14:5060;transport=udp</a>&gt;</div>
<div>X-Siemens-Call-Type: ST-insecure</div>
<div>Date: Tue, 01 Mar 2011 11:59:41 GMT</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 298</div>
<div><br></div>
<div>v=0</div>
<div>o=- 1298983958 1298983958 IN IP4 194.138.116.14</div>
<div>s=RG8700 0lx0486 0x00500101</div>
<div>c=IN IP4 194.138.116.14</div>
<div>t=0 0</div>
<div>a=sendrecv</div>
<div>a=pmft:T38</div>
<div>m=audio 30172 RTP/AVP 8 96</div>
<div>a=rtpmap:8 PCMA/8000</div>
<div>a=rtpmap:96 telephone-event/8000</div>
<div>a=fmtp:96 0-15</div>
<div>a=sqn:0</div>
<div>a=cdsc:1 image udptl t38</div>
<div>a=ptime:20</div>
<div>a=sendrecv</div>
<div><br></div>
<div>--end msg--</div>
<div>12:59:41.266&nbsp; &nbsp; &nbsp; &nbsp; pjsua.c&nbsp; Call 2 state changed to EARLY (180 Ringing)</div>
<div>12:59:41.267&nbsp; strm0x607c774&nbsp; VAD temporarily disabled</div>
<div>12:59:41.267&nbsp; strm0x607c774&nbsp; Encoder stream started</div>
<div>12:59:41.267&nbsp; strm0x607c774&nbsp; Decoder stream started</div>
<div>12:59:41.268&nbsp; pjsua_media.c&nbsp; Media updates, stream #0: PCMA (sendrecv)</div>
<div>12:59:41.268 &nbsp; conference.c&nbsp; Port 3 (<a href="sip:0030343467824 <at> 194.138.116.14">sip:0030343467824 <at> 194.138.116.14</a>) transmitting to port 0 (iPhone IO device)</div>
<div>12:59:41.268 &nbsp; conference.c&nbsp; Port 0 (iPhone IO device) transmitting to port 3 (<a href="sip:0030343467824 <at> 194.138.116.14">sip:0030343467824 <at> 194.138.116.14</a>)</div>
<div>12:59:41.269&nbsp; &nbsp; &nbsp; &nbsp; pjsua.c&nbsp; Media for call 2 is active</div>
<div>12:59:41.895&nbsp; strm0x607c774&nbsp; VAD re-enabled</div>
<div>12:59:43.374 &nbsp; pjsua_core.c&nbsp; RX 1043 bytes Response msg 200/INVITE/cseq=12614 (rdata0x608bc14) from UDP 194.138.116.14:5060:</div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 79.197.201.235:45242;branch=z9hG4bKPjmzNISAbsYUjezdOw.BKUNkjrNo.O4ILm;rport=45242</div>
<div>From: <a href="sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7">sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7</a>
</div>
<div>To: &lt;<a href="sip:0030343467824 <at> 194.138.116.14">sip:0030343467824 <at> 194.138.116.14</a>&gt;;tag=SEC11-a7aa8c0-1e7aa8c0-1-1aJ2ZxBVt57P</div>
<div>Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV</div>
<div>CSeq: 12614 INVITE</div>
<div>Contact: &lt;<a href="sip:0030343467824 <at> 194.138.116.14:5060;transport=udp">sip:0030343467824 <at> 194.138.116.14:5060;transport=udp</a>&gt;</div>
<div>X-Siemens-Call-Type: ST-insecure</div>
<div>Accept-Language: en;q=0.0</div>
<div>Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO</div>
<div>P-Asserted-Identity: &lt;<a href="sip:30343467824 <at> 192.168.122.38">sip:30343467824 <at> 192.168.122.38</a>&gt;</div>
<div>Require: timer</div>
<div>Session-Expires: 1800;refresher=uas</div>
<div>Supported: timer</div>
<div>Date: Tue, 01 Mar 2011 11:59:43 GMT</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 298</div>
<div><br></div>
<div>v=0</div>
<div>o=- 1298983958 1298983958 IN IP4 194.138.116.14</div>
<div>s=RG8700 0lx0486 0x00500101</div>
<div>c=IN IP4 194.138.116.14</div>
<div>t=0 0</div>
<div>a=sendrecv</div>
<div>a=pmft:T38</div>
<div>m=audio 30172 RTP/AVP 8 96</div>
<div>a=rtpmap:8 PCMA/8000</div>
<div>a=rtpmap:96 telephone-event/8000</div>
<div>a=fmtp:96 0-15</div>
<div>a=sqn:0</div>
<div>a=cdsc:1 image udptl t38</div>
<div>a=ptime:20</div>
<div>a=sendrecv</div>
<div><span class="Apple-style-span"><br></span></div>
<div>12:59:43.375&nbsp; &nbsp; &nbsp; &nbsp; pjsua.c&nbsp; Call 2 state changed to CONNECTING</div>
<div>12:59:43.375 &nbsp; inv0xc819864&nbsp; SDP negotiation done, message body is ignored</div>
<div>12:59:43.376 &nbsp; pjsua_core.c&nbsp; TX 417 bytes Request msg ACK/cseq=12614 (tdta0x608f000) to UDP 194.138.116.14:5060:</div>
<div>ACK <a href="sip:0030343467824 <at> 194.138.116.14:5060;transport=udp">sip:0030343467824 <at> 194.138.116.14:5060;transport=udp</a> SIP/2.0</div>
<div>Via: SIP/2.0/UDP 79.197.201.235:45242;rport;branch=z9hG4bKPjkB9AY5itqdhldn-V0u5WAIFw-QKfoDsx</div>
<div>Max-Forwards: 70</div>
<div>From: <a href="sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7">sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7</a>
</div>
<div>To: <a href="sip:0030343467824 <at> 194.138.116.14;tag=SEC11-a7aa8c0-1e7aa8c0-1-1aJ2ZxBVt57P">sip:0030343467824 <at> 194.138.116.14;tag=SEC11-a7aa8c0-1e7aa8c0-1-1aJ2ZxBVt57P</a>
</div>
<div>Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV</div>
<div>CSeq: 12614 ACK</div>
<div>Content-Length:&nbsp; 0</div>
<div><br></div>
<div>12:59:43.376&nbsp; &nbsp; &nbsp; &nbsp; pjsua.c&nbsp; Call 2 state changed to CONFIRMED</div>
<div><div>&nbsp;&nbsp;</div></div>
<div>12:59:47.167 &nbsp; pjsua_core.c&nbsp; TX 452 bytes Request msg BYE/cseq=12615 (tdta0x6092000) to UDP 194.138.116.14:5060:</div>
<div>BYE <a href="sip:0030343467824 <at> 194.138.116.14:5060;transport=udp">sip:0030343467824 <at> 194.138.116.14:5060;transport=udp</a> SIP/2.0</div>
<div>Via: SIP/2.0/UDP 79.197.201.235:45242;rport;branch=z9hG4bKPjdO8FSLaxJeuskHWBsewdOUMLnLGhrSqy</div>
<div>Max-Forwards: 70</div>
<div>From: <a href="sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7">sip:49894423416532 <at> 194.138.116.14;tag=CDHD2RMC6798AKPBcYS0jD.iBtGU3Js7</a>
</div>
<div>To: <a href="sip:0030343467824 <at> 194.138.116.14;tag=SEC11-a7aa8c0-1e7aa8c0-1-1aJ2ZxBVt57P">sip:0030343467824 <at> 194.138.116.14;tag=SEC11-a7aa8c0-1e7aa8c0-1-1aJ2ZxBVt57P</a>
</div>
<div>Call-ID: BFqpf6jm0YGzozKajY3MjDYGtDJQsoWV</div>
<div>CSeq: 12615 BYE</div>
<div>User-Agent: SIP-Phone v0.1/darwin</div>
<div>Content-Length:&nbsp; 0</div>
<div><div><div><div><div><br></div></div></div></div></div>
<div><div><div><div><div>I think the line "SDP negotiation done, message body is ignored" should tell me whats wrong but i cant see the problem..&nbsp;</div></div></div></div></div>
<div><div><div><div><div>can anybody say me whats the problem?</div></div></div></div></div>
<div><div><div><div><div>im grateful for every help i can get&nbsp;</div></div></div></div></div>
<div>
<div><div><div><div></div></div></div></div>
<div><div><div><div><div><div>best regards Matthias&nbsp;</div></div></div></div></div></div>
</div>
</div></div>
</div>
Permalink | Reply | 2 comments |
headers
Philippe Leuba | 1 Mar 2011 14:49
Favicon

Other bug in sip_regc

Hi,

 

I have found another bug in the sip_regc layer.

 

If you send an unregister command, the tsx_callback set the regc->expires to 0.

 

Then when you send a register command, the tsx_callback start the refresh timer to 5 secs instead of the value of the Expires received in the 200 OK (this is due to the regc->expires being 0). The problem occurs only once.

 

Best regards

 

Philippe Leuba

 

From: Philippe Leuba [mailto:philippe.leuba <at> eyepmedia.com]
Sent: lundi 28 février 2011 16:06
To: 'pjsip <at> lists.pjsip.org'
Subject: RE: Bug in sip_regc

 

Hi,

 

I have found what I think is a bug in the sip_regc layer.

 

When you encounter an error (different from 401/407/423), the current_op variable is not reset to REGC_IDLE, preventing to others REGISTER requests (sip_regc_send checks for current_op being REGC_IDLE).

 

This occurs for example when your AOR is already registered for the maximum number of times allowed, you receive a 403 error, but you can not send an unregister_all command.

 

Philippe Leuba

 

<div>

<div class="Section1">

<p class="MsoNormal"><span>Hi,<p></p></span></p>

<p class="MsoNormal"><span><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-GB">I have found another bug in
the sip_regc layer.<p></p></span></p>

<p class="MsoNormal"><span lang="EN-GB"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-GB">If you send an unregister
command, the tsx_callback set the regc-&gt;expires to 0.<p></p></span></p>

<p class="MsoNormal"><span lang="EN-GB"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-GB">Then when you send a
register command, the tsx_callback start the refresh timer to 5 secs instead of
the value of the Expires received in the 200 OK (this is due to the
regc-&gt;expires being 0). The problem occurs only once.<p></p></span></p>

<p class="MsoNormal"><span lang="EN-GB"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-GB">Best regards<p></p></span></p>

<p class="MsoNormal"><span lang="EN-GB"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-GB">Philippe Leuba<p></p></span></p>

<p class="MsoNormal"><span lang="EN-GB"><p>&nbsp;</p></span></p>

<div>

<div class="MsoNormal" align="center"><span lang="EN-US">

</span></div>

<p class="MsoNormal"><span lang="EN-US">From:</span><span lang="EN-US">
Philippe Leuba [mailto:philippe.leuba <at> eyepmedia.com] <br><span>Sent:</span> lundi 28 f&eacute;vrier 2011 16:06<br><span>To:</span> 'pjsip <at> lists.pjsip.org'<br><span>Subject:</span> RE: Bug in sip_regc</span><span lang="EN-US"><p></p></span></p>

</div>

<p class="MsoNormal"><span><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span>Hi,<p></p></span></p>

<p class="MsoNormal"><span><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-GB">I have found what I think
is a bug in the sip_regc layer.<p></p></span></p>

<p class="MsoNormal"><span lang="EN-GB"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-GB">When you encounter an
error (different from 401/407/423), the current_op variable is not reset to
REGC_IDLE, preventing to others REGISTER requests (sip_regc_send checks for
current_op being REGC_IDLE).<p></p></span></p>

<p class="MsoNormal"><span lang="EN-GB"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-GB">This occurs for example
when your AOR is already registered for the maximum number of times allowed,
you receive a 403 error, but you can not send an unregister_all command.<p></p></span></p>

<p class="MsoNormal"><span lang="EN-GB"><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span lang="EN-GB">Philippe Leuba<p></p></span></p>

<p class="MsoNormal"><span lang="EN-GB"><p>&nbsp;</p></span></p>

</div>

</div>
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headers
Jorge | 1 Mar 2011 17:57
Picon

pjsip 2.0

Hello:

As far as I know pjsip 2.0 supports ffmpeg but when I try to start vstream_util I get :

vstreamutil: ../src/pjmedia-codec/ffmpeg_codecs.c:214: pjmedia_codec_ffmpeg_init: Assertion `ci->info.dir != PJMEDIA_DIR_ENCODING_DECODING' failed.

That I get when is initializing all all the codecs in :

pjmedia_codec_ffmpeg_init(pjmedia_vid_codec_mgr *mgr, pj_pool_factory *pf)

It is trying to initialize all the codecs also a encoder-decoder one that is PJMEDIA_DIR_ENCODING_DECODING.

How can I select the codecs I want to initialize ?

Regards

Jorge.




<div><p>Hello:<br><br>As far as I know pjsip 2.0 supports ffmpeg but when I try to start vstream_util I get :<br><br>vstreamutil: ../src/pjmedia-codec/ffmpeg_codecs.c:214: pjmedia_codec_ffmpeg_init: Assertion `ci-&gt;info.dir != PJMEDIA_DIR_ENCODING_DECODING' failed.<br><br>That I get when is initializing all all the codecs in :<br><br>pjmedia_codec_ffmpeg_init(pjmedia_vid_codec_mgr *mgr, pj_pool_factory *pf)<br><br>It is trying to initialize all the codecs also a encoder-decoder one that is PJMEDIA_DIR_ENCODING_DECODING. <br><br>How can I select the codecs I want to initialize ?<br><br>Regards<br><br>Jorge.<br><br><br><br><br></p></div>
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headers
Saúl Ibarra Corretgé | 1 Mar 2011 19:03
Favicon
Gravatar

Re: SDP negotiation done, message body is ignored

On 03/01/2011 01:22 PM, matthias <at> infinatic.de wrote:
> Hello,
> i`ve developed a application with pjsua and tested them with a local
> sip-server and everything works fine but if i try to make a call with a
> extern sip server only my conversational partner can hear me, and no rtp
> is sended by him. In debug messages i can see:
> *
> *
[snip]

In your trace I see some private IP addresses. Is your server doing NAT 
traversal for you? Otherwise packets will not reach the other end.

--

-- 
Saúl Ibarra Corretgé
AG Projects
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headers
Thomas Martin | 1 Mar 2011 22:45
Picon
Picon

compilation error building 1.8.10 (latest trunk) for iphone simulator SDK 4.2

Hello Everybody,

today I have svn-updated my 1.8.10 and successfully recompiled for the device.

However, compiling for the simulator creates the following error:

/Developer/Platforms/iPhoneSimulator.platform/Developer/usr/bin/gcc-4.0 -c -Wall
-DPJ_AUTOCONF=1 -O2 -Wno-unused-label -DPJ_SDK_NAME="\"iPhoneSimulator4.2.sdk\""  -isysroot
/Developer/Platforms/iPhoneSimulator.platform/Developer/SDKs/iPhoneSimulator4.2.sdk   
-I../include \
		-ooutput/pjlib-arm-apple-darwin9/os_info_iphone.o \
		../src/pj/os_info_iphone.m 
In file included from
../src/pj/os_info_iphone.m:23:
/Developer/Platforms/iPhoneSimulator.platform/Developer/SDKs/iPhoneSimulator4.2.sdk/System/Library/Frameworks/UIKit.framework/Headers/UIDevice.h:33:
error: syntax error before ‘}’ token
make[2]: *** [output/pjlib-arm-apple-darwin9/os_info_iphone.o] Error 1
make[1]: *** [pjlib] Error 2
make: *** [all] Error 1

after:

export DEVPATH=/Developer/Platforms/iPhoneSimulator.platform/Developer
export CC=/Developer/Platforms/iPhoneSimulator.platform/Developer/usr/bin/gcc-4.0

./configure-iphone

make dep
make clean
make

The config_site.h looks like this:

#define PJ_CONFIG_IPHONE 1
#include <pj/config_site_sample.h>

#undef __MAC_OS_X_VERSION_MIN_REQUIRED
#undef __MAC_OS_X_VERSION_MAX_ALLOWED
#define __MAC_OS_X_VERSION_MIN_REQUIRED __MAC_OS_X_VERSION_10_4
#define __MAC_OS_X_VERSION_MAX_ALLOWED __MAC_OS_X_VERSION_10_5

Looking forward to a hint ...

Thanks in advance!

-Thomas
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headers
Ming | 2 Mar 2011 03:45
Favicon

Re: compilation error building 1.8.10 (latest trunk) for iphone simulator SDK 4.2

Hi Thomas,

Thanks for the report. We just fixed this in r3433.

Best regards,
Ming

On Wed, Mar 2, 2011 at 5:45 AM, Thomas Martin <tmemail <at> gmx.de> wrote:
> Hello Everybody,
>
> today I have svn-updated my 1.8.10 and successfully recompiled for the device.
>
> However, compiling for the simulator creates the following error:
>
> /Developer/Platforms/iPhoneSimulator.platform/Developer/usr/bin/gcc-4.0 -c -Wall
-DPJ_AUTOCONF=1 -O2 -Wno-unused-label -DPJ_SDK_NAME="\"iPhoneSimulator4.2.sdk\""  -isysroot
/Developer/Platforms/iPhoneSimulator.platform/Developer/SDKs/iPhoneSimulator4.2.sdk  
 -I../include \
>                -ooutput/pjlib-arm-apple-darwin9/os_info_iphone.o \
>                ../src/pj/os_info_iphone.m
> In file included from ../src/pj/os_info_iphone.m:23:
>
/Developer/Platforms/iPhoneSimulator.platform/Developer/SDKs/iPhoneSimulator4.2.sdk/System/Library/Frameworks/UIKit.framework/Headers/UIDevice.h:33:
error: syntax error before ‘}’ token
> make[2]: *** [output/pjlib-arm-apple-darwin9/os_info_iphone.o] Error 1
> make[1]: *** [pjlib] Error 2
> make: *** [all] Error 1
>
> after:
>
> export DEVPATH=/Developer/Platforms/iPhoneSimulator.platform/Developer
> export CC=/Developer/Platforms/iPhoneSimulator.platform/Developer/usr/bin/gcc-4.0
>
> ./configure-iphone
>
> make dep
> make clean
> make
>
> The config_site.h looks like this:
>
> #define PJ_CONFIG_IPHONE 1
> #include <pj/config_site_sample.h>
>
> #undef __MAC_OS_X_VERSION_MIN_REQUIRED
> #undef __MAC_OS_X_VERSION_MAX_ALLOWED
> #define __MAC_OS_X_VERSION_MIN_REQUIRED __MAC_OS_X_VERSION_10_4
> #define __MAC_OS_X_VERSION_MAX_ALLOWED __MAC_OS_X_VERSION_10_5
>
> Looking forward to a hint ...
>
> Thanks in advance!
>
> -Thomas
>
>
> _______________________________________________
> 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
>
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Gmane