Re: [pjsip] Video and Audio not working when using SRTP in iOS
Stranger. Try hook SRTP in Wireshark.
18 июля 2019 г., в 17:07, Александр Клейменов a.kleymenov@encry.ru написал(а):
I am getting same without "add to registrar uri and call peer uri ;transport=tls;lr «
Simple try
18 июля 2019 г., в 17:05, Anuran Barman <anuranbarman@gmail.com mailto:anuranbarman@gmail.com> написал(а):
"Try add to registrar uri and call peer uri ;transport=tls;lr "
I already have this in my call uri and register uri. Call is established but video and audio not working.
On Thu, Jul 18, 2019 at 7:33 PM Александр Клейменов <a.kleymenov@encry.ru mailto:a.kleymenov@encry.ru> wrote:
Without this setting I am getting call without voice too.
Try add to registrar uri and call peer uri ;transport=tls;lr
Me help that, but ONLY in release. In debug no voice over TLS
18 июля 2019 г., в 16:58, Anuran Barman <anuranbarman@gmail.com mailto:anuranbarman@gmail.com> написал(а):
I am able to make the call. Just the video and audio is not working. TLS setting is correct only as you can see in the logs, it's communicating via TLS only. and those certificates are optional I guess as Linphone is working fine without those certificates.
On Thu, Jul 18, 2019 at 7:25 PM Александр Клейменов <a.kleymenov@encry.ru mailto:a.kleymenov@encry.ru> wrote:
When creating acc with TLS I am setting
val tlsCfg = TlsConfig()
tlsCfg.certFile = certPath
tlsCfg.privKeyFile = certPath
tlsCfg.verifyServer = false
tlsCfg.method = pjsip_ssl_method.PJSIP_TLSV1_METHOD
accCfg.mediaConfig.srtpUse = pjmedia_srtp_use.PJMEDIA_SRTP_MANDATORY
accCfg.mediaConfig.srtpSecureSignaling = 1
accCfg.mediaConfig.transportConfig.tlsConfig = tlsCfg
Adding ;transport=tls;lr to registrar uri and uri when making call
Hope this help you
18 июля 2019 г., в 16:42, Anuran Barman <anuranbarman@gmail.com mailto:anuranbarman@gmail.com> написал(а):
More on that is using two instances of linphone I am able to make the video call fine. If i turn of SRTP and use RTP in PJSIP everything works fine. Only when using SRTP it's creating the problem. The way I am configuring is like below:
ua_cfg.use_srtp = PJMEDIA_SRTP_MANDATORY;
ua_cfg.srtp_secure_signaling = PJSUA_DEFAULT_SRTP_SECURE_SIGNALING;
pjsua_srtp_opt srtp_opt;
pjsua_srtp_opt_default(&srtp_opt);
ua_cfg.srtp_opt = srtp_opt;
ua_cfg.srtp_optional_dup_offer = PJ_TRUE;
It looks like it also does not work in android. This is the exact problem I am facing in ios. Please help regarding this. What can be the isssue?
Android Similar Problem:https://stackoverflow.com/questions/56031734/how-to-enable-srtp-with-pjsip-in-android https://stackoverflow.com/questions/56031734/how-to-enable-srtp-with-pjsip-in-android
On Thu, Jul 18, 2019 at 5:26 PM Anuran Barman <anuranbarman@gmail.com mailto:anuranbarman@gmail.com> wrote:
I am able to register and get the call. How can I get that if the settings are not correct. Only video and audio is not working.
On Thu, Jul 18, 2019 at 5:24 PM Александр Клейменов <a.kleymenov@encry.ru mailto:a.kleymenov@encry.ru> wrote:
Are you sure in account settings for TLS?
18 июля 2019 г., в 14:27, Anuran Barman <anuranbarman@gmail.com mailto:anuranbarman@gmail.com> написал(а):
Hi, No even in release it is not working. Same problem.
On Thu, Jul 18, 2019 at 4:53 PM Александр Клейменов <a.kleymenov@encry.ru mailto:a.kleymenov@encry.ru> wrote:
Hello!
A have same problem on Android in debug, in release work nice - try release.
Visit our blog: http://blog.pjsip.org http://blog.pjsip.org/
pjsip mailing list
pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org http://blog.pjsip.org/
pjsip mailing list
pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org http://blog.pjsip.org/
pjsip mailing list
pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org http://blog.pjsip.org/
pjsip mailing list
pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org http://blog.pjsip.org/
pjsip mailing list
pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org http://blog.pjsip.org/
pjsip mailing list
pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org http://blog.pjsip.org/
pjsip mailing list
pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org http://blog.pjsip.org/
pjsip mailing list
pjsip@lists.pjsip.org mailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Not working after adding also. TLS does not have any problem I guess. It's
the SRTP which has the problem.
On Thu, Jul 18, 2019 at 7:38 PM Александр Клейменов a.kleymenov@encry.ru
wrote:
I am getting same without "add to registrar uri and call peer uri
;transport=tls;lr «
Simple try
18 июля 2019 г., в 17:05, Anuran Barman anuranbarman@gmail.com
написал(а):
"Try add to registrar uri and call peer uri ;transport=tls;lr "
I already have this in my call uri and register uri. Call is established
but video and audio not working.
On Thu, Jul 18, 2019 at 7:33 PM Александр Клейменов a.kleymenov@encry.ru
wrote:
Without this setting I am getting call without voice too.
Try add to registrar uri and call peer uri ;transport=tls;lr
Me help that, but ONLY in release. In debug no voice over TLS
18 июля 2019 г., в 16:58, Anuran Barman anuranbarman@gmail.com
написал(а):
I am able to make the call. Just the video and audio is not working. TLS
setting is correct only as you can see in the logs, it's communicating via
TLS only. and those certificates are optional I guess as Linphone is
working fine without those certificates.
On Thu, Jul 18, 2019 at 7:25 PM Александр Клейменов a.kleymenov@encry.ru
wrote:
When creating acc with TLS I am setting
val tlsCfg = TlsConfig()
tlsCfg.certFile = certPath
tlsCfg.privKeyFile = certPath
tlsCfg.verifyServer = false
tlsCfg.method = pjsip_ssl_method.PJSIP_TLSV1_METHOD
accCfg.mediaConfig.srtpUse = pjmedia_srtp_use.PJMEDIA_SRTP_MANDATORY
accCfg.mediaConfig.srtpSecureSignaling = 1
accCfg.mediaConfig.transportConfig.tlsConfig = tlsCfg
Adding ;transport=tls;lr to registrar uri and uri when making call
Hope this help you
18 июля 2019 г., в 16:42, Anuran Barman anuranbarman@gmail.com
написал(а):
More on that is using two instances of linphone I am able to make the
video call fine. If i turn of SRTP and use RTP in PJSIP everything works
fine. Only when using SRTP it's creating the problem. The way I am
configuring is like below:
ua_cfg.use_srtp = PJMEDIA_SRTP_MANDATORY;
ua_cfg.srtp_secure_signaling = PJSUA_DEFAULT_SRTP_SECURE_SIGNALING;
pjsua_srtp_opt srtp_opt;
pjsua_srtp_opt_default(&srtp_opt);
ua_cfg.srtp_opt = srtp_opt;
ua_cfg.srtp_optional_dup_offer = PJ_TRUE;
It looks like it also does not work in android. This is the exact
problem I am facing in ios. Please help regarding this. What can be the
isssue?
Android Similar Problem:
https://stackoverflow.com/questions/56031734/how-to-enable-srtp-with-pjsip-in-android
On Thu, Jul 18, 2019 at 5:26 PM Anuran Barman anuranbarman@gmail.com
wrote:
I am able to register and get the call. How can I get that if the
settings are not correct. Only video and audio is not working.
On Thu, Jul 18, 2019 at 5:24 PM Александр Клейменов <
a.kleymenov@encry.ru> wrote:
Are you sure in account settings for TLS?
18 июля 2019 г., в 14:27, Anuran Barman anuranbarman@gmail.com
написал(а):
Hi, No even in release it is not working. Same problem.
On Thu, Jul 18, 2019 at 4:53 PM Александр Клейменов <
a.kleymenov@encry.ru> wrote:
Hello!
A have same problem on Android in debug, in release work nice - try
release.
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
It seems like key exchange has some problem. I changed key exchange to
DTLS_SRTP and made the keying_count to 1. Now video and audio still doesn't
work but the error is different. Error is like below when a call is
established :
2019-07-19 10:16:00.808371+0530 AnuranRealSIPOBJ[6456:1497494] configuring
audio session..
10:16:00.936 pjsua_call.c !Making call with acc #0 to
sips:1001@anuran.barman.com:5061;transport=tls;lr
10:16:00.936 pjsua_aud.c .Set sound device: capture=-1,
playback=-2
10:16:00.936 pjsua_aud.c ..Opening sound device (speaker + mic)
PCM@16000/1/20ms
10:16:00.936 coreaudio_dev.c ...Using VoiceProcessingIO audio unit
10:16:01.173 coreaudio_dev.c ...core audio stream started
10:16:01.177 os_core_unix.c Info: possibly re-registering existing
thread
10:16:01.192 pjsua_media.c .Call 1: initializing media..
10:16:01.192 pjsua_media.c ..RTP socket reachable at
103.78.19.128:4020
10:16:01.192 pjsua_media.c ..RTCP socket reachable at
103.78.19.128:4021
10:16:01.192 pjsua_media.c ..RTP socket reachable at
103.78.19.128:4022
10:16:01.192 pjsua_media.c ..RTCP socket reachable at
103.78.19.128:4023
10:16:01.192 pjsua_media.c ..Media index 0 selected for audio
call 1
10:16:01.192 srtp0x102027000 .SRTP uses keying method DTLS-SRTP
10:16:01.192 srtp0x102008800 .SRTP uses keying method DTLS-SRTP
10:16:01.194 pjsua_core.c ....TX 1719 bytes Request msg
INVITE/cseq=8827 (tdta0x1020782a8) to TLS 103.154.197.129:5061:
INVITE sips:1001@anuran.barman.com:5061;transport=tls;lr SIP/2.0
Via: SIP/2.0/TLS 103.78.19.128:59729
;rport;branch=z9hG4bKPjRJN62m2C2M0XaVWPZrS8uY6IjJhURDys;alias
Max-Forwards: 70
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com
Contact: sips:3001@103.78.19.128:59729;transport=TLS;ob
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8827 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
User-Agent: AnuranRealSIPObj
Content-Type: application/sdp
Content-Length: 1040
v=0
o=- 3772500361 3772500361 IN IP4 103.78.19.128
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 4020 UDP/TLS/RTP/SAVP 0 8 98 97 3 99 104 9 96
c=IN IP4 103.78.19.128
b=TIAS:64000
a=rtcp:4021 IN IP4 103.78.19.128
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ssrc:1370242855 cname:1ad4ff0215b1ea11
a=setup:actpass
a=fingerprint:SHA-256
81:46:F2:5C:3C:12:CD:8A:F3:69:A7:92:17:8F:4F:0A:6C:5F:B8:92:3E:05:8A:8A:89:64:A7:4B:DE:72:82:EE
m=video 4022 UDP/TLS/RTP/SAVP 97
c=IN IP4 103.78.19.128
b=TIAS:256000
a=rtcp:4023 IN IP4 103.78.19.128
a=sendrecv
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1
a=ssrc:57033557 cname:1ad4ff0215b1ea11
a=setup:actpass
a=fingerprint:SHA-256
81:46:F2:5C:3C:12:CD:8A:F3:69:A7:92:17:8F:4F:0A:6C:5F:B8:92:3E:05:8A:8A:89:64:A7:4B:DE:72:82:EE
--end msg--
10:16:01.194 APP .......anuran callState 1 state=CALLING
2019-07-19 10:16:01.194625+0530 AnuranRealSIPOBJ[6456:1497494] call made
successfully from 0 with callID 1
2019-07-19 10:16:01.194647+0530 AnuranRealSIPOBJ[6456:1497494] call is
active
10:16:01.402 pjsua_core.c .RX 541 bytes Response msg
407/INVITE/cseq=8827 (rdata0x10203ad48) from TLS 103.154.197.129:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TLS 103.78.19.128:59729
;rport=59729;branch=z9hG4bKPjRJN62m2C2M0XaVWPZrS8uY6IjJhURDys;alias;received=103.78.19.128
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com;tag=fd456a75ca976fba91e6fa5a990d400a.2fd7
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8827 INVITE
Proxy-Authenticate: Digest realm="anuran.barman.com",
nonce="XTFMNV0xSwlTgASAoUIm/BU3vCdvEa+l"
Server: kamailio (4.4.7 (x86_64/linux))
Content-Length: 0
--end msg--
10:16:01.402 pjsua_core.c ..TX 411 bytes Request msg
ACK/cseq=8827 (tdta0x10703dca8) to TLS 103.154.197.129:5061:
ACK sips:1001@anuran.barman.com:5061;transport=tls;lr SIP/2.0
Via: SIP/2.0/TLS 103.78.19.128:59729
;rport;branch=z9hG4bKPjRJN62m2C2M0XaVWPZrS8uY6IjJhURDys;alias
Max-Forwards: 70
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com;tag=fd456a75ca976fba91e6fa5a990d400a.2fd7
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8827 ACK
Content-Length: 0
--end msg--
10:16:01.403 pjsua_core.c .......TX 1937 bytes Request msg
INVITE/cseq=8828 (tdta0x1020782a8) to TLS 103.154.197.129:5061:
INVITE sips:1001@anuran.barman.com:5061;transport=tls;lr SIP/2.0
Via: SIP/2.0/TLS 103.78.19.128:59729
;rport;branch=z9hG4bKPjTb8lyPMfaohy6r7mURlXulhHTlEXZWaW;alias
Max-Forwards: 70
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com
Contact: sips:3001@103.78.19.128:59729;transport=TLS;ob
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8828 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
User-Agent: AnuranRealSIPObj
Proxy-Authorization: Digest username="3001", realm="anuran.barman.com",
nonce="XTFMNV0xSwlTgASAoUIm/BU3vCdvEa+l",
uri="sips:1001@anuran.barman.com:5061;transport=tls;lr",
response="03c793a98347eeb7b28cef73e5bcef44"
Content-Type: application/sdp
Content-Length: 1040
v=0
o=- 3772500361 3772500361 IN IP4 103.78.19.128
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 4020 UDP/TLS/RTP/SAVP 0 8 98 97 3 99 104 9 96
c=IN IP4 103.78.19.128
b=TIAS:64000
a=rtcp:4021 IN IP4 103.78.19.128
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ssrc:1370242855 cname:1ad4ff0215b1ea11
a=setup:actpass
a=fingerprint:SHA-256
81:46:F2:5C:3C:12:CD:8A:F3:69:A7:92:17:8F:4F:0A:6C:5F:B8:92:3E:05:8A:8A:89:64:A7:4B:DE:72:82:EE
m=video 4022 UDP/TLS/RTP/SAVP 97
c=IN IP4 103.78.19.128
b=TIAS:256000
a=rtcp:4023 IN IP4 103.78.19.128
a=sendrecv
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1
a=ssrc:57033557 cname:1ad4ff0215b1ea11
a=setup:actpass
a=fingerprint:SHA-256
81:46:F2:5C:3C:12:CD:8A:F3:69:A7:92:17:8F:4F:0A:6C:5F:B8:92:3E:05:8A:8A:89:64:A7:4B:DE:72:82:EE
--end msg--
10:16:01.607 pjsua_core.c .RX 411 bytes Response msg
100/INVITE/cseq=8828 (rdata0x10203ad48) from TLS 103.154.197.129:5061:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/TLS 103.78.19.128:59729
;rport=59729;branch=z9hG4bKPjTb8lyPMfaohy6r7mURlXulhHTlEXZWaW;alias;received=103.78.19.128
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8828 INVITE
Server: kamailio (4.4.7 (x86_64/linux))
Content-Length: 0
--end msg--
10:16:02.017 pjsua_core.c .RX 543 bytes Response msg
180/INVITE/cseq=8828 (rdata0x10203ad48) from TLS 103.154.197.129:5061:
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 103.78.19.128:59729
;received=103.78.19.128;rport=59729;branch=z9hG4bKPjTb8lyPMfaohy6r7mURlXulhHTlEXZWaW;alias
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com;tag=eVT1Z-~
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8828 INVITE
User-Agent: LinphoneiOS/4.1 (Anuran’s iPhone) LinphoneSDK/4.2
(belle-sip/1.6.3)
Supported: replaces, outbound, gruu
Record-route: sips:103.154.197.129:5061;transport=tls;lr;nat=yes
Content-Length: 0
--end msg--
10:16:02.017 APP .....anuran callState 1 state=EARLY
10:16:05.191 pjsua_core.c .RX 1691 bytes Response msg
200/INVITE/cseq=8828 (rdata0x10203ad48) from TLS 103.154.197.129:5061:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 103.78.19.128:59729
;received=103.78.19.128;rport=59729;branch=z9hG4bKPjTb8lyPMfaohy6r7mURlXulhHTlEXZWaW;alias
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com;tag=eVT1Z-~
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8828 INVITE
User-Agent: LinphoneiOS/4.1 (Anuran’s iPhone) LinphoneSDK/4.2
(belle-sip/1.6.3)
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
Contact: <sips:1001@103.78.19.128:57552
;transport=tls;alias=103.78.19.128~57552~3>;expires=3600;received="sip:103.78.19.128:57552
;transport=tls";+sip.instance="urn:uuid:35030849-844f-0077-b310-a34b30fe5950";+org.linphone.specs="lime"
Content-Type: application/sdp
Content-Length: 792
Record-route: sips:103.154.197.129:5061;transport=tls;lr;nat=yes
v=0
o=1001 785 1476 IN IP4 103.154.197.129
s=Talk
c=IN IP4 103.154.197.129
t=0 0
m=audio 21194 UDP/TLS/RTP/SAVP 0 8 98 97 3 104 9 96
a=rtpmap:98 speex/16000
a=fmtp:98 vbr=on
a=rtpmap:97 speex/8000
a=fmtp:97 vbr=on
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:96 telephone-event/8000
a=setup:active
a=fingerprint:SHA-256
DF:DC:F3:85:33:4D:31:82:54:AA:38:7A:E1:8C:71:F6:08:6E:7D:EE:7D:C4:0D:90:1D:EA:1B:C0:7A:88:71:43
a=ssrc:2645697335 cname:sips:3001@anuran.barman.com
m=video 29742 UDP/TLS/RTP/SAVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F
a=setup:active
a=fingerprint:SHA-256
DF:DC:F3:85:33:4D:31:82:54:AA:38:7A:E1:8C:71:F6:08:6E:7D:EE:7D:C4:0D:90:1D:EA:1B:C0:7A:88:71:43
a=ssrc:2554319350 cname:sips:3001@anuran.barman.com
a=nortpproxy:yes
--end msg--
10:16:05.191 APP .....anuran callState 1
state=CONNECTING
10:16:05.192 inv0x1020350a8 ....SDP negotiation done: Success
10:16:05.192 pjsua_media.c .....Call 1: updating media..
10:16:05.192 pjsua_media.c .......Media stream call01:0 is
destroyed
10:16:05.192 pjsua_aud.c ......Audio channel update..
10:16:05.192 strm0x102841228 .......VAD temporarily disabled
10:16:05.192 strm0x102841228 .......Error sending RTCP: SRTP
parameters negotiation still in progress (PJMEDIA_SRTP_EKEYNOTREADY)
10:16:05.192 strm0x102841228 .......Encoder stream started
10:16:05.192 strm0x102841228 .......Decoder stream started
10:16:05.192 pjsua_media.c ......Audio updated, stream #0: PCMU
(sendrecv)
10:16:05.192 pjsua_media.c .......Media stream call01:1 is
destroyed
10:16:05.192 pjsua_vid.c ......Video channel update..
10:16:05.193 strm0x102841228 Error sending RTP: SRTP parameters
negotiation still in progress (PJMEDIA_SRTP_EKEYNOTREADY)
10:16:05.201 vstenc0x102846428 .......Error sending RTCP: SRTP
parameters negotiation still in progress (PJMEDIA_SRTP_EKEYNOTREADY)
10:16:05.201 vstenc0x102846428 .......Encoder stream started
10:16:05.201 vstdec0x102846428 .......Decoder stream started
10:16:05.201 pjsua_vid.c .......Setting up RX..
10:16:05.201 pjsua_vid.c ........Creating video window:
type=stream, cap_id=-1, rend_id=0
10:16:05.201 vid_port.c .........Opening device OpenGL
renderer [OpenGL] for render: format=I420, size=656x656 @22:1 fps
10:16:05.217 ios_opengl_dev.c .........iOS OpenGL ES renderer
successfully created
10:16:05.217 vid_port.c .........Device OpenGL renderer
[OpenGL] opened: format=BGRA, size=656x656 @22:1 fps
10:16:05.218 vid_conf.c .........Added port 0 (OpenGL renderer)
10:16:05.218 pjsua_vid.c .........stream window id 0 created
for cap_dev=-1 rend_dev=0
10:16:05.218 pjsua_vid.c .........Window 0 created
10:16:05.219 vid_conf.c ........Added port 1
(vstdec0x102846428)
10:16:05.219 vid_conf.c ........Port 1 (vstdec0x102846428)
transmitting to port 0 (OpenGL renderer)
10:16:05.219 ios_opengl_dev.c ........Starting ios opengl stream
10:16:05.219 pjsua_vid.c .......Setting up TX..
10:16:05.219 vid_conf.c ........Added port 2
(vstenc0x102846428)
10:16:05.219 pjsua_vid.c ........Creating video window:
type=preview, cap_id=2, rend_id=0
10:16:05.219 vid_port.c .........Opening device Front Camera
[AVF] for capture: format=I420, size=352x288 @15:1 fps
10:16:05.226 vid_util.c .........Orientation converter
created: 288x352 to 236x288, maintain aspect ratio=yes
10:16:05.226 vid_port.c .........Device Front Camera [AVF]
opened: format=I420, size=352x288 @15:1 fps
10:16:05.235 vid_conf.c .........Added port 3 (Front Camera)
10:16:05.236 darwin_dev.m !.........Native preview initialized
10:16:05.236 pjsua_vid.c .........Preview window id 1 created
for cap_dev 2, using built-in preview!
10:16:05.236 pjsua_vid.c .........Window 1 created
10:16:05.236 vid_conf.c ........Port 3 (Front Camera)
transmitting to port 2 (vstenc0x102846428)
10:16:05.236 darwin_dev.m ........Starting Darwin video stream
10:16:05.625 pjsua_media.c ......Video updated, stream #1: H264
(sendrecv)
10:16:05.625 APP .....Call 1 media 0 [type=audio],
status is Active
10:16:05.625 pjsua_aud.c .....Conf connect: 1 --> 0
10:16:05.625 conference.c ......Port 1
(sips:1001@anuran.barman.com:5061;transport=tls;lr) transmitting to port 0
(iPhone IO device)
10:16:05.625 pjsua_aud.c .....Conf connect: 0 --> 1
10:16:05.625 conference.c ......Port 0 (iPhone IO device)
transmitting to port 1 (sips:1001@anuran.barman.com:5061;transport=tls;lr)
10:16:05.625 APP .....Call 1 media 1 [type=video],
status is Active
arrange window executing
arrange SRTP has
arrange window has video
arrange window wid id 0
10:16:05.632 pjsua_core.c .....TX 463 bytes Request msg
ACK/cseq=8828 (tdta0x10701f0a8) to TLS 103.154.197.129:5061:
ACK sips:1001@103.78.19.128:57552;transport=tls;alias=103.78.19.128~57552~3
SIP/2.0
Via: SIP/2.0/TLS 103.78.19.128:59729
;rport;branch=z9hG4bKPjwpo-sT3Aq603TT-iLwXMhraDOcmWypUD;alias
Max-Forwards: 70
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com;tag=eVT1Z-~
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8828 ACK
Route: sips:103.154.197.129:5061;transport=tls;lr;nat=yes
Content-Length: 0
--end msg--
10:16:05.632 APP .....anuran callState 1 state=CONFIRMED
10:16:05.635 pjsua_call.c .Call 1 sending UPDATE for updating
media session to use only one codec
10:16:05.635 srtp0x102027000 .SRTP uses keying method DTLS-SRTP
10:16:05.635 srtp0x102008800 .SRTP uses keying method DTLS-SRTP
10:16:05.637 pjsua_core.c ....TX 1483 bytes Request msg
UPDATE/cseq=8829 (tdta0x1070234a8) to TLS 103.154.197.129:5061:
UPDATE sips:1001@103.78.19.128:57552;transport=tls;alias=103.78.19.128~57552~3
SIP/2.0
Via: SIP/2.0/TLS 103.78.19.128:59729
;rport;branch=z9hG4bKPjpfwVnPmwEaSJupuSF0Ia4e6dasub7h0E;alias
Max-Forwards: 70
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com;tag=eVT1Z-~
Contact: sips:3001@103.78.19.128:59729;transport=TLS;ob
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8829 UPDATE
Route: sips:103.154.197.129:5061;transport=tls;lr;nat=yes
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 838
v=0
o=- 3772500361 3772500362 IN IP4 103.78.19.128
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 4020 UDP/TLS/RTP/SAVP 0 96
c=IN IP4 103.78.19.128
b=TIAS:64000
a=rtcp:4021 IN IP4 103.78.19.128
a=ssrc:1370242855 cname:1ad4ff0215b1ea11
a=setup:passive
a=fingerprint:SHA-256
81:46:F2:5C:3C:12:CD:8A:F3:69:A7:92:17:8F:4F:0A:6C:5F:B8:92:3E:05:8A:8A:89:64:A7:4B:DE:72:82:EE
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=sendrecv
m=video 4022 UDP/TLS/RTP/SAVP 97
c=IN IP4 103.78.19.128
b=TIAS:256000
a=rtcp:4023 IN IP4 103.78.19.128
a=ssrc:57033557 cname:1ad4ff0215b1ea11
a=setup:passive
a=fingerprint:SHA-256
81:46:F2:5C:3C:12:CD:8A:F3:69:A7:92:17:8F:4F:0A:6C:5F:B8:92:3E:05:8A:8A:89:64:A7:4B:DE:72:82:EE
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42e01e; packetization-mode=1
a=sendrecv
--end msg--
10:16:05.638 vstenc0x102846428 !Error sending RTP: SRTP parameters
negotiation still in progress (PJMEDIA_SRTP_EKEYNOTREADY)
10:16:05.807 pjsua_core.c .RX 1691 bytes Response msg
200/INVITE/cseq=8828 (rdata0x10203ad48) from TLS 103.154.197.129:5061:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 103.78.19.128:59729
;received=103.78.19.128;rport=59729;branch=z9hG4bKPjTb8lyPMfaohy6r7mURlXulhHTlEXZWaW;alias
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com;tag=eVT1Z-~
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8828 INVITE
User-Agent: LinphoneiOS/4.1 (Anuran’s iPhone) LinphoneSDK/4.2
(belle-sip/1.6.3)
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
Contact: <sips:1001@103.78.19.128:57552
;transport=tls;alias=103.78.19.128~57552~3>;expires=3600;received="sip:103.78.19.128:57552
;transport=tls";+sip.instance="urn:uuid:35030849-844f-0077-b310-a34b30fe5950";+org.linphone.specs="lime"
Content-Type: application/sdp
Content-Length: 792
Record-route: sips:103.154.197.129:5061;transport=tls;lr;nat=yes
v=0
o=1001 785 1476 IN IP4 103.154.197.129
s=Talk
c=IN IP4 103.154.197.129
t=0 0
m=audio 21194 UDP/TLS/RTP/SAVP 0 8 98 97 3 104 9 96
a=rtpmap:98 speex/16000
a=fmtp:98 vbr=on
a=rtpmap:97 speex/8000
a=fmtp:97 vbr=on
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:96 telephone-event/8000
a=setup:active
a=fingerprint:SHA-256
DF:DC:F3:85:33:4D:31:82:54:AA:38:7A:E1:8C:71:F6:08:6E:7D:EE:7D:C4:0D:90:1D:EA:1B:C0:7A:88:71:43
a=ssrc:2645697335 cname:sips:3001@anuran.barman.com
m=video 29742 UDP/TLS/RTP/SAVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F
a=setup:active
a=fingerprint:SHA-256
DF:DC:F3:85:33:4D:31:82:54:AA:38:7A:E1:8C:71:F6:08:6E:7D:EE:7D:C4:0D:90:1D:EA:1B:C0:7A:88:71:43
a=ssrc:2554319350 cname:sips:3001@anuran.barman.com
a=nortpproxy:yes
--end msg--
10:16:05.808 pjsua_core.c ...TX 463 bytes Request msg
ACK/cseq=8828 (tdta0x10701f0a8) to TLS 103.154.197.129:5061:
ACK sips:1001@103.78.19.128:57552;transport=tls;alias=103.78.19.128~57552~3
SIP/2.0
Via: SIP/2.0/TLS 103.78.19.128:59729
;rport;branch=z9hG4bKPjwpo-sT3Aq603TT-iLwXMhraDOcmWypUD;alias
Max-Forwards: 70
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com;tag=eVT1Z-~
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8828 ACK
Route: sips:103.154.197.129:5061;transport=tls;lr;nat=yes
Content-Length: 0
--end msg--
10:16:05.819 strm0x102841228 !VAD re-enabled
10:16:06.217 pjsua_core.c .RX 1208 bytes Response msg
200/UPDATE/cseq=8829 (rdata0x10203ad48) from TLS 103.154.197.129:5061:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 103.78.19.128:59729
;received=103.78.19.128;rport=59729;branch=z9hG4bKPjpfwVnPmwEaSJupuSF0Ia4e6dasub7h0E;alias
From: sips:3001@anuran.barman.com;tag=WB-EU0FxkNkS-0L8TVyCOQWDXvBWKTtR
To: sips:1001@anuran.barman.com;tag=eVT1Z-~
Call-ID: OKgzM2NQq-SvACV3vtXk.PrzAyzQIeDM
CSeq: 8829 UPDATE
User-Agent: LinphoneiOS/4.1 (Anuran’s iPhone) LinphoneSDK/4.2
(belle-sip/1.6.3)
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
Contact: <sips:1001@103.78.19.128:57552
;transport=tls;alias=103.78.19.128~57552~3>;expires=3600;received="sip:103.78.19.128:57552
;transport=tls";+sip.instance="urn:uuid:35030849-844f-0077-b310-a34b30fe5950";+org.linphone.specs="lime"
Content-Type: application/sdp
Content-Length: 377
v=0
o=1001 785 1478 IN IP4 103.154.197.129
s=Talk
c=IN IP4 103.154.197.129
t=0 0
m=audio 21194 UDP/TLS/RTP/SAVP 0 96
a=rtpmap:96 telephone-event/8000
a=ssrc:2645697335 cname:sips:3001@anuran.barman.com
m=video 29742 UDP/TLS/RTP/SAVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F
a=ssrc:2554319350 cname:sips:3001@anuran.barman.com
a=nortpproxy:yes
--end msg--
10:16:06.217 inv0x1020350a8 ....SDP negotiation done: Success
10:16:06.217 pjsua_media.c .....Call 1: updating media..
10:16:06.217 pjsua_media.c ......Call 1: stream #0 (audio)
unchanged.
10:16:06.218 pjsua_media.c ......pjmedia_transport_media_start()
failed for call_id 1 media 0: SRTP SDP contains ambigue answer
(PJMEDIA_SRTP_ESDPAMBIGUEANS)
10:16:06.218 pjsua_media.c .......Media stream call01:0 is
destroyed
10:16:06.218 pjsua_media.c ......Error updating media call01:0:
SRTP SDP contains ambigue answer (PJMEDIA_SRTP_ESDPAMBIGUEANS)
10:16:06.218 pjsua_media.c ......Call 1: stream #1 (video)
unchanged.
10:16:06.218 pjsua_media.c ......pjmedia_transport_media_start()
failed for call_id 1 media 1: SRTP SDP contains ambigue answer
(PJMEDIA_SRTP_ESDPAMBIGUEANS)
10:16:06.218 pjsua_vid.c .......Stopping video stream..
10:16:06.218 vid_conf.c ........Port 3 (Front Camera) stop
transmitting to port 2 (vstenc0x102846428)
10:16:06.218 vid_conf.c ........Removed port 2
(vstenc0x102846428)
10:16:06.218 vid_conf.c ........Port 1 (vstdec0x102846428)
stop transmitting to port 0 (OpenGL renderer)
10:16:06.218 vid_conf.c ........Removed port 1
(vstdec0x102846428)
10:16:06.219 pjsua_vid.c ........Window 1: destroying..
10:16:06.219 vid_conf.c .........Removed port 3 (Front Camera)
10:16:06.219 darwin_dev.m .........Stopping Darwin video stream
10:16:06.299 vid_port.c .........Closing Front Camera..
10:16:06.303 ios_opengl_dev.c ........Stopping ios opengl stream
10:16:06.303 pjsua_vid.c !........Window 0: destroying..
10:16:06.303 vid_conf.c .........Removed port 0 (OpenGL
renderer)
10:16:06.304 ios_opengl_dev.c .........Stopping ios opengl stream
10:16:06.304 vid_port.c .........Closing OpenGL renderer..
10:16:06.310 pjsua_media.c .......Media stream call01:1 is
destroyed
10:16:06.310 pjsua_media.c ......Error updating media call01:1:
SRTP SDP contains ambigue answer (PJMEDIA_SRTP_ESDPAMBIGUEANS)
10:16:06.310 APP .....Call 1 media 0 [type=audio],
status is Error
10:16:06.310 APP .....Call 1 media 1 [type=video],
status is Error
On Thu, Jul 18, 2019 at 7:41 PM Александр Клейменов a.kleymenov@encry.ru
wrote:
Stranger. Try hook SRTP in Wireshark.
18 июля 2019 г., в 17:07, Александр Клейменов a.kleymenov@encry.ru
написал(а):
I am getting same without "add to registrar uri and call peer uri
;transport=tls;lr «
Simple try
18 июля 2019 г., в 17:05, Anuran Barman anuranbarman@gmail.com
написал(а):
"Try add to registrar uri and call peer uri ;transport=tls;lr "
I already have this in my call uri and register uri. Call is established
but video and audio not working.
On Thu, Jul 18, 2019 at 7:33 PM Александр Клейменов a.kleymenov@encry.ru
wrote:
Without this setting I am getting call without voice too.
Try add to registrar uri and call peer uri ;transport=tls;lr
Me help that, but ONLY in release. In debug no voice over TLS
18 июля 2019 г., в 16:58, Anuran Barman anuranbarman@gmail.com
написал(а):
I am able to make the call. Just the video and audio is not working. TLS
setting is correct only as you can see in the logs, it's communicating via
TLS only. and those certificates are optional I guess as Linphone is
working fine without those certificates.
On Thu, Jul 18, 2019 at 7:25 PM Александр Клейменов a.kleymenov@encry.ru
wrote:
When creating acc with TLS I am setting
val tlsCfg = TlsConfig()
tlsCfg.certFile = certPath
tlsCfg.privKeyFile = certPath
tlsCfg.verifyServer = false
tlsCfg.method = pjsip_ssl_method.PJSIP_TLSV1_METHOD
accCfg.mediaConfig.srtpUse = pjmedia_srtp_use.PJMEDIA_SRTP_MANDATORY
accCfg.mediaConfig.srtpSecureSignaling = 1
accCfg.mediaConfig.transportConfig.tlsConfig = tlsCfg
Adding ;transport=tls;lr to registrar uri and uri when making call
Hope this help you
18 июля 2019 г., в 16:42, Anuran Barman anuranbarman@gmail.com
написал(а):
More on that is using two instances of linphone I am able to make the
video call fine. If i turn of SRTP and use RTP in PJSIP everything works
fine. Only when using SRTP it's creating the problem. The way I am
configuring is like below:
ua_cfg.use_srtp = PJMEDIA_SRTP_MANDATORY;
ua_cfg.srtp_secure_signaling = PJSUA_DEFAULT_SRTP_SECURE_SIGNALING;
pjsua_srtp_opt srtp_opt;
pjsua_srtp_opt_default(&srtp_opt);
ua_cfg.srtp_opt = srtp_opt;
ua_cfg.srtp_optional_dup_offer = PJ_TRUE;
It looks like it also does not work in android. This is the exact
problem I am facing in ios. Please help regarding this. What can be the
isssue?
Android Similar Problem:
https://stackoverflow.com/questions/56031734/how-to-enable-srtp-with-pjsip-in-android
On Thu, Jul 18, 2019 at 5:26 PM Anuran Barman anuranbarman@gmail.com
wrote:
I am able to register and get the call. How can I get that if the
settings are not correct. Only video and audio is not working.
On Thu, Jul 18, 2019 at 5:24 PM Александр Клейменов <
a.kleymenov@encry.ru> wrote:
Are you sure in account settings for TLS?
18 июля 2019 г., в 14:27, Anuran Barman anuranbarman@gmail.com
написал(а):
Hi, No even in release it is not working. Same problem.
On Thu, Jul 18, 2019 at 4:53 PM Александр Клейменов <
a.kleymenov@encry.ru> wrote:
Hello!
A have same problem on Android in debug, in release work nice - try
release.
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org