Hi Franz,
I have read this doc but still cannot figure the problem out.
My code sample, log and packet capture are attached.
This sample can make the call and bind the socket to the particular interface (enp1s0 or enp2s0) but only if I use one account.
When I add the 2nd account, the caller sends the INVITE message but the called receives nothing.
When I add the 2nd account, the caller sends the INVITE message but the called receives nothing.
Can you please point to me which part I'm wrong?
Best Regards,
===============================================================
Quy Bui Danh (Mr.)
===============================================================
Quy Bui Danh (Mr.)
Vào Th 2, 24 thg 2, 2020 vào lúc 19:35 Skale Franz <franz.skale@xxxxxxxxxxxxxxxxxxx> đã viết:
Again,
when reading the docs carefully, you have to configure the pjsua_transport_config structure accordingly:
Reference:
https://www.pjsip.org/pjsip/docs/html/structpjsua__transport__config.htm#a76cdb153b60fbbdbfd36ced9494bea0a
Check the docs section for bound_addr and public_addr.
Best regards
Franz
Von: pjsip <pjsip-bounces@xxxxxxxxxxxxxxx> im Auftrag von Bui Quy <quybd@xxxxxxxxxxxx>
Gesendet: Montag, 24. Februar 2020 12:07
An: pjsip list
Betreff: Re: PJSUA with two NICs
Hi Franz,
Thank you for your reply.
Currently, I face the problem like below:
PC1 and PC2 have two different NICS are enp1s0 and enp2s0.
PC1 call to PC2.
In the PC2, I can bound the local account (pjsua_acc_id) to enp1s0 or enp2s0, then the PC2 will respond with the corresponding interface:
pjsua_transport_config udp_cfg;
pjsua_transport_config_default(&udp_cfg);
udp_cfg.port = port;
pjsua_transport_id transport_id;
pjsua_transport_create(PJSIP_TRANSPORT_UDP, &udp_cfg, &transport_id);
pjsua_acc_add_local(transport_id, PJ_TRUE, &acc);
pjsua_acc_config acc_cfg;
pjsua_acc_config_default(&acc_cfg);
pjsua_acc_get_config(acc, pool, &acc_cfg);
acc_cfg.rtp_cfg.public_addr = pj_str(my_ip);
acc_cfg.rtp_cfg.bound_addr = pj_str(my_ip);
pjsua_acc_modify(acc, &acc_cfg);
But after added the 1st account, if I add the 2nd local account into the PC B, with different udp_cfg.port, the PC2 always responds with the interface which was bounded by the 2nd account, even if the 2nd account listens on a different port.
pjsua_transport_config udp_cfg_2;
pjsua_transport_config_default(&udp_cfg_2);
udp_cfg.port = port_2;
pjsua_transport_id transport_id_2;
pjsua_transport_create(PJSIP_TRANSPORT_UDP, &udp_cfg_2, &transport_id_2);
pjsua_acc_add_local(transport_id_2, PJ_TRUE, &acc_2);
pjsua_acc_config acc_cfg_2;
pjsua_acc_config_default(&acc_cfg_2);
pjsua_acc_get_config(acc_2, pool, &acc_cfg_2);
acc_cfg.rtp_cfg.public_addr = pj_str(my_ip_2);
acc_cfg.rtp_cfg.bound_addr = pj_str(my_ip_2);
pjsua_acc_modify(acc_2, &acc_cfg_2);
My goal is to make PC1 able to call to PC2 with two separate calls on the different interfaces.
Best Regards,
===============================================================
Quy Bui Danh (Mr.)
Vào Th 6, 21 thg 2, 2020 vào lúc 17:32 Skale Franz <franz.skale@xxxxxxxxxxxxxxxxxxx> đã viết:
Hi,
you have to bind the specific IP address to the specific interface:
https://www.pjsip.org/pjsip/docs/html/structpjsua__transport__config.htm#a672e097722cbc1d4308d1bcd7bb4d911
Best regards
Franz
Von: pjsip <pjsip-bounces@xxxxxxxxxxxxxxx> im Auftrag von Bui Quy <quybd@xxxxxxxxxxxx>
Gesendet: Freitag, 21. Februar 2020 11:13
An: pjsip@xxxxxxxxxxxxxxx
Betreff: PJSUA with two NICs
Hi,
I would like to set up a system like below:
(172.36.100.11) (172.36.100.12)
enp1s0 ----------------------------------- enp1s0
PC 1 ------ | |------------ PC 2
enp2s0 ----------------------------------- enp2s0
(172.36.101.11) (172.36.101.12)
I can make a call from PC1 to PC2 via enp2s0 interface (172.36.101.11), but all the RTP packets are always going through enp1s0 (172.36.100.11) not enp2s0 (172.36.101.11).
How can I make those RTP packets go through the enp2s0 interface?
I follow this site to config two default gateway:
https://www.thomas-krenn.com/en/wiki/Two_Default_Gateways_on_One_System
On PC1:
route -n
Destination Gateway Genmask Flags Metric Ref Use Iface
0.0.0.0 172.36.100.1 0.0.0.0 UG 0 0 0 enp1s0
169.254.0.0 0.0.0.0 255.255.0.0 U 1000 0 0 enp1s0
172.36.100.0 0.0.0.0 255.255.255.0 U 0 0 0 enp1s0
172.36.101.0 0.0.0.0 255.255.255.0 U 0 0 0 enp2s0
ip route list table rt2
default via 172.36.101.1 dev enp2s0
172.36.101.0/24 dev enp2s0 scope link src 172.36.101.12
ip rule show
0: from all lookup local
32764: from all to 172.36.101.12/24 lookup rt2
32765: from 172.36.101.12/24 lookup rt2
32766: from all lookup main
32767: from all lookup default
When I make a call from PC1 to PC2 via enp2s0, this is the INVITE message:
Internet Protocol Version 4, Src: 172.36.101.11, Dst: 172.36.101.12
User Datagram Protocol, Src Port: 4321, Dst Port: 1234
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:172.36.101.12:1234 SIP/2.0
Message Header
Via: SIP/2.0/UDP 172.36.100.11:4321;rport;branch=z9hG4bKPjXPf3YS3e4e6gZ0m4z.oDyeO0KNHwknAE
Max-Forwards: 70
From: <sip:172.36.100.11>;tag=Fd18S4-WSwPB3bxA7.QstNy.GomS4lSO
To: sip:172.36.101.246
Contact: <sip:172.36.101.11:4321;ob>
Call-ID: MTqhGS7DmAfQxEvZSlH9gpf3be.UycnQ
[Generated Call-ID: MTqhGS7DmAfQxEvZSlH9gpf3be.UycnQ]
CSeq: 16789 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 476
Message Body
As you can observe, my enp1s0 address (172.36.100.11) is in the Via and From headers.
I can force those headers to use enp2s0 address (172.36.101.11) by adding this line at the end of pj_gethostip() function:
PJ_DEF(pj_status_t) pj_gethostip(int af, pj_sockaddr *addr)
{
....
....
pj_sockaddr_copy_addr(addr, &cand_addr[selected_cand]);
TRACE_((THIS_FILE, "Candidate %s selected",
pj_sockaddr_print(addr, strip, sizeof(strip), 0)));
}
addr->ipv4.sin_addr.s_addr = inet_addr("172.36.101.11");
return PJ_SUCCESS;
}
But even after that, all the RTP packets still going through enp1s0.
Any advice would be most welcome!
Best Regards,
===============================================================
Quy Bui Danh (Mr.)
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04:46:47.509 os_core_unix.c !pjlib 2.5.5 for POSIX initialized 04:46:47.518 alsa_dev.c Added sound device default:CARD=PCH 04:46:47.518 alsa_dev.c Added sound device sysdefault:CARD=PCH 04:46:47.519 alsa_dev.c Added sound device hw:CARD=PCH,DEV=0 04:46:47.519 alsa_dev.c Added sound device hw:CARD=PCH,DEV=1 04:46:47.526 alsa_dev.c Added sound device hw:CARD=PCH,DEV=3 04:46:47.526 alsa_dev.c Added sound device hw:CARD=PCH,DEV=7 04:46:47.527 alsa_dev.c Added sound device hw:CARD=PCH,DEV=8 04:46:47.527 alsa_dev.c Added sound device hw:CARD=PCH,DEV=9 04:46:47.527 alsa_dev.c Added sound device hw:CARD=PCH,DEV=10 04:46:47.528 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=0 04:46:47.528 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=1 04:46:47.528 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=3 04:46:47.529 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=7 04:46:47.529 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=8 04:46:47.529 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=9 04:46:47.530 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=10 04:46:47.530 alsa_dev.c ALSA driver found 16 devices 04:46:47.530 alsa_dev.c ALSA initialized 04:46:47.532 pjlib select() I/O Queue created (0x1d103c8) 04:46:47.535 sip_endpoint.c .Creating endpoint instance... 04:46:47.536 pjlib .select() I/O Queue created (0x1d7fc98) 04:46:47.536 sip_endpoint.c .Module "mod-msg-print" registered 04:46:47.536 sip_transport. .Transport manager created. 04:46:47.536 pjsua_core.c .PJSUA state changed: NULL --> CREATED 04:46:47.536 sip_endpoint.c .Module "mod-pjsua-log" registered 04:46:47.536 sip_endpoint.c .Module "mod-tsx-layer" registered 04:46:47.536 sip_endpoint.c .Module "mod-stateful-util" registered 04:46:47.536 sip_endpoint.c .Module "mod-ua" registered 04:46:47.536 sip_endpoint.c .Module "mod-100rel" registered 04:46:47.536 sip_endpoint.c .Module "mod-pjsua" registered 04:46:47.537 sip_endpoint.c .Module "mod-invite" registered 04:46:47.539 pjlib ..select() I/O Queue created (0x1dbf208) 04:46:47.539 conference.c ..Creating conference bridge with 254 ports 04:46:47.539 Master/sound ..Using delay buffer with WSOLA. 04:46:47.547 sip_endpoint.c .Module "mod-evsub" registered 04:46:47.547 sip_endpoint.c .Module "mod-presence" registered 04:46:47.547 evsub.c .Event pkg "presence" registered by mod-presence 04:46:47.547 sip_endpoint.c .Module "mod-mwi" registered 04:46:47.547 evsub.c .Event pkg "message-summary" registered by mod-mwi 04:46:47.547 sip_endpoint.c .Module "mod-refer" registered 04:46:47.547 evsub.c .Event pkg "refer" registered by mod-refer 04:46:47.547 sip_endpoint.c .Module "mod-pjsua-pres" registered 04:46:47.547 sip_endpoint.c .Module "mod-pjsua-im" registered 04:46:47.547 sip_endpoint.c .Module "mod-pjsua-options" registered 04:46:47.547 pjsua_core.c .1 SIP worker threads created 04:46:47.547 pjsua_core.c .pjsua version 2.5.5 for Linux-4.15.0.72/x86_64/glibc-2.23 initialized 04:46:47.547 pjsua_core.c .PJSUA state changed: CREATED --> INIT 04:46:47.547 pjsua_aud.c Setting null sound device.. 04:46:47.547 pjsua_aud.c .Opening null sound device.. 04:46:47.548 pjsua_core.c SIP UDP socket reachable at 172.36.100.12:4321 04:46:47.548 udp0x1e0eb40 SIP UDP transport started, published address is 172.36.100.12:4321 04:46:47.548 pjsua_acc.c Adding account: id=<sip:172.36.100.12:4321> 04:46:47.548 pjsua_acc.c .Account <sip:172.36.100.12:4321> added with id 0 04:46:47.548 pjsua_core.c SIP UDP socket reachable at 172.36.101.12:4322 04:46:47.548 udp0x1e12700 SIP UDP transport started, published address is 172.36.101.12:4322 04:46:47.548 pjsua_acc.c Adding account: id=<sip:172.36.101.12:4322> 04:46:47.548 pjsua_acc.c .Account <sip:172.36.101.12:4322> added with id 1 04:46:47.548 pjsua_core.c PJSUA state changed: INIT --> STARTING 04:46:47.548 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 04:46:47.548 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING m 04:46:50.564 src/sample_pho Call to sip:if_1@172.36.100.19:1234 04:46:50.564 pjsua_call.c Making call with acc #0 to sip:if_1@172.36.100.19:1234 04:46:50.564 dlg0x1e16da8 .UAC dialog created 04:46:50.564 dlg0x1e16da8 ..Session count inc to 2 by mod-pjsua 04:46:50.564 pjsua_media.c .Call 0: initializing media.. 04:46:50.565 pjsua_media.c ..RTP socket reachable at 172.36.100.12:4000 04:46:50.565 pjsua_media.c ..RTCP socket reachable at 172.36.100.12:4001 04:46:50.565 pjsua_media.c ..Media index 0 selected for audio call 0 04:46:50.565 dlg0x1e16da8 ..Session count dec to 2 by mod-pjsua 04:46:50.565 dlg0x1e16da8 .Module mod-invite added as dialog usage, data=0x1e1cd98 04:46:50.565 dlg0x1e16da8 ..Session count inc to 4 by mod-invite 04:46:50.565 dlg0x1e16da8 .Module mod-100rel added as dialog usage, data=0x1e1e968 04:46:50.565 dlg0x1e16da8 .100rel module attached 04:46:50.565 inv0x1e16da8 .UAC invite session created for dialog dlg0x1e16da8 04:46:50.565 endpoint .Request msg INVITE/cseq=1386 (tdta0x1e1ec10) created. 04:46:50.565 inv0x1e16da8 ..Sending Request msg INVITE/cseq=1386 (tdta0x1e1ec10) 04:46:50.565 dlg0x1e16da8 ...Sending Request msg INVITE/cseq=1386 (tdta0x1e1ec10) 04:46:50.565 tsx0x1e21c28 ....Transaction created for Request msg INVITE/cseq=1385 (tdta0x1e1ec10) 04:46:50.565 tsx0x1e21c28 ...Sending Request msg INVITE/cseq=1385 (tdta0x1e1ec10) in state Null 04:46:50.565 sip_resolve.c ....Target '172.36.100.19:1234' type=Unspecified resolved to '172.36.100.19:1234' type=UDP (UDP transport) 04:46:50.565 pjsua_core.c ....TX 1059 bytes Request msg INVITE/cseq=1385 (tdta0x1e1ec10) to UDP 172.36.100.19:1234: INVITE sip:if_1@172.36.100.19:1234 SIP/2.0 Via: SIP/2.0/UDP 172.36.100.12:4322;rport;branch=z9hG4bKPj.FeB5TaPC4pQ2RhberdeFvSCkMbyBW1b Max-Forwards: 70 From: <sip:172.36.100.12>;tag=RxX-HZ2cciFgi5pu6jgOZnafha1aKEK6 To: sip:if_1@172.36.100.19 Contact: <sip:172.36.100.12:4322;ob> Call-ID: ikX-UmTyX0du8aQYEkcVFZji9ZyaV9c- CSeq: 1385 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 476 v=0 o=- 3791594810 3791594810 IN IP4 172.36.100.12 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 172.36.100.12 b=TIAS:64000 a=rtcp:4001 IN IP4 172.36.100.12 a=sendrecv 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=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 --end msg-- 04:46:50.566 tsx0x1e21c28 ....State changed from Null to Calling, event=TX_MSG 04:46:50.566 dlg0x1e16da8 .....Transaction tsx0x1e21c28 state changed to Calling 04:46:50.566 src/sample_pho .......Call 0 state changed to CALLING 04:46:51.067 tsx0x1e21c28 !Retransmit timer event 04:46:51.067 tsx0x1e21c28 .Retransmiting Request msg INVITE/cseq=1385 (tdta0x1e1ec10), count=0, restart?=1 04:46:51.067 pjsua_core.c .TX 1059 bytes Request msg INVITE/cseq=1385 (tdta0x1e1ec10) to UDP 172.36.100.19:1234: INVITE sip:if_1@172.36.100.19:1234 SIP/2.0 Via: SIP/2.0/UDP 172.36.100.12:4322;rport;branch=z9hG4bKPj.FeB5TaPC4pQ2RhberdeFvSCkMbyBW1b Max-Forwards: 70 From: <sip:172.36.100.12>;tag=RxX-HZ2cciFgi5pu6jgOZnafha1aKEK6 To: sip:if_1@172.36.100.19 Contact: <sip:172.36.100.12:4322;ob> Call-ID: ikX-UmTyX0du8aQYEkcVFZji9ZyaV9c- CSeq: 1385 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 476 v=0 o=- 3791594810 3791594810 IN IP4 172.36.100.12 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 172.36.100.12 b=TIAS:64000 a=rtcp:4001 IN IP4 172.36.100.12 a=sendrecv 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=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 --end msg-- 04:46:52.067 tsx0x1e21c28 Retransmit timer event 04:46:52.067 tsx0x1e21c28 .Retransmiting Request msg INVITE/cseq=1385 (tdta0x1e1ec10), count=1, restart?=1 04:46:52.067 pjsua_core.c .TX 1059 bytes Request msg INVITE/cseq=1385 (tdta0x1e1ec10) to UDP 172.36.100.19:1234: INVITE sip:if_1@172.36.100.19:1234 SIP/2.0 Via: SIP/2.0/UDP 172.36.100.12:4322;rport;branch=z9hG4bKPj.FeB5TaPC4pQ2RhberdeFvSCkMbyBW1b Max-Forwards: 70 From: <sip:172.36.100.12>;tag=RxX-HZ2cciFgi5pu6jgOZnafha1aKEK6 To: sip:if_1@172.36.100.19 Contact: <sip:172.36.100.12:4322;ob> Call-ID: ikX-UmTyX0du8aQYEkcVFZji9ZyaV9c- CSeq: 1385 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 476 v=0 o=- 3791594810 3791594810 IN IP4 172.36.100.12 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 c=IN IP4 172.36.100.12 b=TIAS:64000 a=rtcp:4001 IN IP4 172.36.100.12 a=sendrecv 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=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 --end msg-- 04:46:53.567 tsx0x1e21c28 Timeout timer event 04:46:53.567 tsx0x1e21c28 .State changed from Calling to Terminated, event=TIMER 04:46:53.567 dlg0x1e16da8 ..Transaction tsx0x1e21c28 state changed to Terminated 04:46:53.567 src/sample_pho ....Call 0 state changed to DISCONNCTD 04:46:53.567 pjsua_media.c ....Call 0: deinitializing media.. 04:46:53.567 pjsua_media.c .....Call 0: cleaning up provisional media, prov_med_cnt=1, med_cnt=0 04:46:53.567 dlg0x1e16da8 ....Session count dec to 1 by mod-invite 04:46:53.567 dlg0x1e16da8 ...Dialog destroyed 04:46:53.567 tsx0x1e21c28 Timeout timer event 04:46:53.567 tsx0x1e21c28 .State changed from Terminated to Destroyed, event=TIMER 04:46:53.567 tdta0x1e1ec10 ..Destroying txdata Request msg INVITE/cseq=1385 (tdta0x1e1ec10) 04:46:53.567 tsx0x1e21c28 Transaction destroyed!
Attachment:
sip.pcapng
Description: Binary data
04:45:56.312 os_core_unix.c !pjlib 2.5.5 for POSIX initialized 04:45:56.329 alsa_dev.c Added sound device default:CARD=PCH 04:45:56.329 alsa_dev.c Added sound device sysdefault:CARD=PCH 04:45:56.329 alsa_dev.c Added sound device hw:CARD=PCH,DEV=0 04:45:56.330 alsa_dev.c Added sound device hw:CARD=PCH,DEV=1 04:45:56.337 alsa_dev.c Added sound device hw:CARD=PCH,DEV=3 04:45:56.338 alsa_dev.c Added sound device hw:CARD=PCH,DEV=7 04:45:56.338 alsa_dev.c Added sound device hw:CARD=PCH,DEV=8 04:45:56.338 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=0 04:45:56.339 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=1 04:45:56.339 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=3 04:45:56.339 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=7 04:45:56.340 alsa_dev.c Added sound device plughw:CARD=PCH,DEV=8 04:45:56.340 alsa_dev.c ALSA driver found 12 devices 04:45:56.340 alsa_dev.c ALSA initialized 04:45:56.342 pjlib select() I/O Queue created (0x27cb3c8) 04:45:56.344 sip_endpoint.c .Creating endpoint instance... 04:45:56.345 pjlib .select() I/O Queue created (0x283b0b8) 04:45:56.345 sip_endpoint.c .Module "mod-msg-print" registered 04:45:56.345 sip_transport. .Transport manager created. 04:45:56.345 pjsua_core.c .PJSUA state changed: NULL --> CREATED 04:45:56.345 sip_endpoint.c .Module "mod-pjsua-log" registered 04:45:56.345 sip_endpoint.c .Module "mod-tsx-layer" registered 04:45:56.345 sip_endpoint.c .Module "mod-stateful-util" registered 04:45:56.345 sip_endpoint.c .Module "mod-ua" registered 04:45:56.345 sip_endpoint.c .Module "mod-100rel" registered 04:45:56.345 sip_endpoint.c .Module "mod-pjsua" registered 04:45:56.346 sip_endpoint.c .Module "mod-invite" registered 04:45:56.348 pjlib ..select() I/O Queue created (0x287a628) 04:45:56.348 conference.c ..Creating conference bridge with 254 ports 04:45:56.348 Master/sound ..Using delay buffer with WSOLA. 04:45:56.354 sip_endpoint.c .Module "mod-evsub" registered 04:45:56.354 sip_endpoint.c .Module "mod-presence" registered 04:45:56.354 evsub.c .Event pkg "presence" registered by mod-presence 04:45:56.354 sip_endpoint.c .Module "mod-mwi" registered 04:45:56.354 evsub.c .Event pkg "message-summary" registered by mod-mwi 04:45:56.354 sip_endpoint.c .Module "mod-refer" registered 04:45:56.354 evsub.c .Event pkg "refer" registered by mod-refer 04:45:56.354 sip_endpoint.c .Module "mod-pjsua-pres" registered 04:45:56.354 sip_endpoint.c .Module "mod-pjsua-im" registered 04:45:56.354 sip_endpoint.c .Module "mod-pjsua-options" registered 04:45:56.354 pjsua_core.c .1 SIP worker threads created 04:45:56.354 pjsua_core.c .pjsua version 2.5.5 for Linux-4.4.0.21/x86_64/glibc-2.23 initialized 04:45:56.354 pjsua_core.c .PJSUA state changed: CREATED --> INIT 04:45:56.354 pjsua_aud.c Setting null sound device.. 04:45:56.354 pjsua_aud.c .Opening null sound device.. 04:45:56.354 pjsua_core.c SIP UDP socket reachable at 172.36.100.19:1234 04:45:56.354 udp0x28c9e10 SIP UDP transport started, published address is 172.36.100.19:1234 04:45:56.354 pjsua_acc.c Adding account: id=<sip:172.36.100.19:1234> 04:45:56.354 pjsua_acc.c .Account <sip:172.36.100.19:1234> added with id 0 04:45:56.355 pjsua_core.c SIP UDP socket reachable at 172.36.101.19:1235 04:45:56.355 udp0x28cd9d0 SIP UDP transport started, published address is 172.36.101.19:1235 04:45:56.355 pjsua_acc.c Adding account: id=<sip:172.36.101.19:1235> 04:45:56.355 pjsua_acc.c .Account <sip:172.36.101.19:1235> added with id 1 04:45:56.355 pjsua_core.c PJSUA state changed: INIT --> STARTING 04:45:56.355 sip_endpoint.c .Module "mod-unsolicited-mwi" registered 04:45:56.355 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
Attachment:
sample_phone.c
Description: Binary data
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